irclog2html for #asterisk on 20061215

00:00.40ManxPowe1EmleyMoor: the CLI is your friend.  Hug it, hold it, buy it flowers.
00:01.11ManxPowe1Also Noop is your friend too
00:01.17EmleyMoorI figure it may be cutting that is the problem
00:01.31ManxPowe1that's why I have the noop after the guts
00:01.35ManxPowe1and the cuts too
00:01.50*** join/#asterisk spammer (i=dp@190.48.132.213)
00:02.02spammerStarted talking in perl on Jueves 14/12/06 20:54:52
00:02.04spammerRoom topic is: No pasting, use http://sial.org/pbot/perl/ or http://erxz.com/pb/ or http://dragon.cbi.tamucc.edu:8080 :: Visit #perlcafe, #perl6 and #pound-perl.pm :: <ew73> VB.NET is all of the fun of enforced privacy OO with all of the power of BASIC :: <GumbyBRAIN> oh really? Rofl
00:02.04spammer#perl [freenode-info] please register your nickname...don't forget to auto-identify! http://freenode.net/faq.shtml#nicksetup
00:02.04spammer20:55<- cjeris has left perl
00:02.06spammerStarted talking in perl on Jueves 14/12/06 20:54:52
00:02.06spammerRoom topic is: No pasting, use http://sial.org/pbot/perl/ or http://erxz.com/pb/ or http://dragon.cbi.tamucc.edu:8080 :: Visit #perlcafe, #perl6 and #pound-perl.pm :: <ew73> VB.NET is all of the fun of enforced privacy OO with all of the power of BASIC :: <GumbyBRAIN> oh really? Rofl
00:02.10spammer#perl [freenode-info] please register your nickname...don't forget to auto-identify! http://freenode.net/faq.shtml#nicksetup
00:02.13spammer20:55<- cjeris has left perl
00:02.14spammerStarted talking in perl on Jueves 14/12/06 20:54:52
00:02.17spammerRoom topic is: No pasting, use http://sial.org/pbot/perl/ or http://erxz.com/pb/ or http://dragon.cbi.tamucc.edu:8080 :: Visit #perlcafe, #perl6 and #pound-perl.pm :: <ew73> VB.NET is all of the fun of enforced privacy OO with all of the power of BASIC :: <GumbyBRAIN> oh really? Rofl
00:02.21spammer#perl [freenode-info] please register your nickname...don't forget to auto-identify! http://freenode.net/faq.shtml#nicksetup
00:02.23spammer20:55<- cjeris has left perl
00:02.25spammerStarted talking in perl on Jueves 14/12/06 20:54:52
00:02.28spammerRoom topic is: No pasting, use http://sial.org/pbot/perl/ or http://erxz.com/pb/ or http://dragon.cbi.tamucc.edu:8080 :: Visit #perlcafe, #perl6 and #pound-perl.pm :: <ew73> VB.NET is all of the fun of enforced privacy OO with all of the power of BASIC :: <GumbyBRAIN> oh really? Rofl
00:02.31spammer#perl [freenode-info] please register your nickname...don't forget to auto-identify! http://freenode.net/faq.shtml#nicksetup
00:02.35spammer20:55<- cjeris has left perl
00:02.37spammerStarted talking in perl on Jueves 14/12/06 20:54:52
00:02.37*** mode/#asterisk [+b *!*@190.48.132.*] by file
00:02.37*** kick/#asterisk [spammer!n=file@asterisk/developer-and-muffin-lover/file] by file (file)
00:03.27DrukenHMEwell, that was fun....
00:03.40backbluei had one orgasm....
00:03.54backbluedam lady spammer
00:04.13*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
00:04.53[hC]ok this is messed.... have a client with a PAP2 and a cisco phone. if they call a LD number from the cisco the call sounds great, if they do it from the PAP2, it sounds terrible and jittery...
00:04.58[hC]What the heck would cause this?
00:05.26DrukenHMEis a local call with the pap2 solid?
00:05.29Qwell[]different codecs?
00:05.36[hC]Druken: yes.
00:05.47*** join/#asterisk borg (n=niklesod@unaffiliated/botxj)
00:05.57DrukenHMEi'd check the codecs between the pap2 and the LD carrier
00:06.27[hC]Druken: tried both ulaw and g729. all of it comes thru me first anyhow, and uses the same codec on the cisco, which sounds fine
00:06.50[hC]the cisco/pap2 go to an in house * server, then iax's to me, then either goes pri for local, or sip for LD to another provider.
00:07.35DrukenHMEthat the only customer with a problem on LD ?
00:08.32[hC]Druken: Yep, and only from this PAP2. all other phones in their office are fine.
00:08.47ManxPowe1[hC]: all other PAP2s too?
00:08.49[hC]Wonder if i can find a newer firmware for the PAP2..
00:09.01[hC]ManxPowe1: all other pap2's exhibit the same problem, yep.
00:09.10ManxPowe1I'll bet the rtp audio packet size is set to somthing other than 20ms
00:09.16ManxPowe1in the pap2
00:09.19[hC]its 0.30
00:09.22[hC]i JUST looked at that.
00:09.32ManxPowe1[hC] well THATS not going to work very well
00:09.38blitzragelol... oh man... gratuitous use of dialplan functions: http://pastebin.ca/279386
00:10.04ManxPowe1[hC] set it to 0.20
00:10.25DrukenHMEwhat is the firmware version on the pap2?
00:10.33*** join/#asterisk dasenjo (n=dasenjo@208.195.215.71)
00:10.39blitzrageor use allow=g729:30
00:10.42[hC]3.1.12(LS)
00:10.49blitzragemight only be in 1.4 though
00:10.49[hC]I definitely didnt change that.
00:10.51ManxPowe1blitzrage: what verison of Asterisk supports that?
00:10.52robl^blitzrage: that's like almost unreadable!! its hideous..  its . . . .  ewwww!
00:10.57blitzrageManxPowe1: 1.2
00:11.13[hC]It was ulaw before, and now ive got it on g729a. It worked better on g729.. i just set him to 0.20 and we'll see whats up.
00:11.17ManxPowe1blitzrage: I really HATE it when people suggest things that only work in a non-release version of Asterisk
00:11.32FarrisGNow even with those ports open, I get one-way audio.
00:11.45blitzrageManxPowe1: the :30?  I think that might be 1.4 actually
00:11.52DrukenHMEone way audio sucks...
00:12.07EmleyMoorIs "CHAN" a reserved variable name?
00:12.07FarrisGDrukenHME: Agreed,  but I can't track down the cause
00:12.07*** join/#asterisk darkmaniac (n=darkmani@bl5-45-184.dsl.telepac.pt)
00:12.11darkmaniachttp://www.gimmickry.org/ <-riddles
00:12.13*** part/#asterisk darkmaniac (n=darkmani@bl5-45-184.dsl.telepac.pt)
00:12.18ManxPowe1FarrisG: When you do a "sip show peers" is the IP address listed the public IP or the private IP?
00:13.24FarrisGManxPowe1: Which one? The only addresses are the external public IP address of my provider, and my internal IP address for the one phone I currently have connected
00:13.35[hC]Well
00:13.36[hC]That fixed it.
00:14.07ManxPowe1FarrisG: canreinvite=no should fix one way audio.
00:14.09EmleyMoorCan I paste the "force voiptalk" part of my dialplan to pastebin for someone to tell me why it sets CHAN to the same as CHANNEL when it should be the bit before the - only?
00:14.14EmleyMoorI will paste anyway...
00:14.14DrukenHMEhc: oops....
00:14.20*** join/#asterisk betatester (n=tester@pool-71-251-229-244.rcmdva.fios.verizon.net)
00:14.23*** join/#asterisk visba_ (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net)
00:14.27blitzragerobl^: yes... not that pretty -- could be better if the dialplan parser let me use CR's
00:14.27FarrisGManxPowe1: Pretty sure I already have that...
00:14.31FarrisGchecking now
00:14.32*** join/#asterisk jm|home (n=jamiem@zen.jamiem.com)
00:14.34[hC]Druken: it must have come that way, nobody here would have dicked with that... how strange.
00:14.51ManxPowe1FarrisG: do you have bindaddr= anywher ein sip.conf?
00:14.54EmleyMoorhttp://pastebin.ca/279397
00:14.59robl^blitzrage: that's something that AEL2 would make prettier
00:15.13blitzragerobl^: yah... if AEL2 let you do something like that
00:15.36EmleyMoorI am testing over Zap/2 because that phone is right by this computer
00:15.41blitzrageI look at enough dialplan stuff now that stuff like that doesn't bother me too much
00:15.43FarrisGManxPowe1: bindaddr=0.0.0.0
00:16.05ManxPowe1EmleyMoor: do a Noop for ${CHANNEL}
00:16.05EmleyMoorIt seems that it is not identifying the fact and is setting CALLERID(num) to the fallback
00:16.26ManxPowe1EmleyMoor: are you SEEING it set CALLERID(num) on the CLI?
00:16.35EmleyMoorYes, I am seeing that
00:16.38*** part/#asterisk visba_ (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net)
00:16.40betatesterFYI: Cisco MWI patch available at http://bugs.digium.com/view.php?id=8575
00:16.59EmleyMoorIt is showing a wrong value for CHAN - Zap/2-1 for example where it should be Zap/2
00:17.26ManxPowe1EmleyMoor: make SURE to do a "show function cut" to make sure the syntax has not changed
00:17.48EmleyMoorI'm using the app but have shown both that and the function
00:17.56EmleyMoorBesides, for TECHNOLOGY, it works
00:19.39EmleyMoorIt seems that the Cut for CHAN is going wrong somehow
00:19.53ManxPowe1pastebin the cli output
00:20.29FarrisGAre there any tell tale signs in logs or console that would let me know how or why I can't hear a caller but they can hear me?
00:20.54ManxPowe1FarrisG: not really as it is almost always either a firewall problem or a NAT problem.
00:20.57JTFarrisG: using NAT?
00:21.02ManxPowe1canreinvite=no will take care of the NAT issues.
00:21.22ManxPowe1FarrisG: without canreinvite=no the phone might try talking directly to the provider
00:21.27EmleyMoorIt's getting the right value but not acting on it
00:21.27FarrisGcanreinvite=no is there. Firewall is wide open for the time being
00:21.51ManxPowe1FarrisG: well run a tcpdump on the server and see where the packets are going
00:22.34JTor rtp debug in * sli
00:22.36JTcli
00:22.40ManxPowe1EmleyMoor: I see the problem!
00:23.07ManxPowe1NOT GotoIf($[${CHAN} = "Zap/1"]?10:7) but GotoIf($["${CHAN}" = "Zap/1"]?10:7)
00:23.32EmleyMoorAh, merci! Diolch yn fawr!
00:23.51ManxPowe1If you use quotes on one side of the comparison then use it on both sides
00:24.38ManxPowe1that info is for ANYONE I've helped
00:24.52wiseoldowlnaftali5 you over here?
00:24.54FarrisGManxPowe1: Ok, so I added my external IP and localnaet to sip_nat.conf, reloaded sip, and now it works. But I'm a little skeptical that it's what fixed it
00:25.13*** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net)
00:25.25JTFarrisG: err, why? NAT and firewalls are typically the only thing that cause SIP oneway audio
00:25.25ManxPowe1FarrisG: I don't know why either since Asterisk has 1 public IP and 1 private IP
00:25.44ManxPowe1unless that is NOT true
00:25.46JTa sip debug may have told you why
00:25.46robin__szJT, well, remove them again and reboot, see of its borked again, then you will kow for sure
00:26.12DrukenHMEanyone here use chanspy?
00:26.18robin__szsorry, FarrisG I meant
00:27.53*** join/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com)
00:27.58*** part/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com)
00:28.44*** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl)
00:29.47orlokhmm...
00:29.57orloki have one of those cheap USB phones
00:30.10orlokplugged it into linuxbox, it appears as a USB audio device, heh
00:30.30orlokcan evenuse the buttons toadjust volume
00:30.37orlokbut cant hear any audio, heh
00:30.40Rawplayerany dutch people here?
00:31.20Rawplayeronly for the dutch people http://www.lidl.nl/nl/home.nsf/pages/c.o.20061214.p.Draadloze_VOIP-telefoon
00:31.31brianI'm not Dutch.
00:31.33Rawplayerits a debranded siemens 450 IP
00:31.35brianBut I'm going to click it anyways.
00:31.40brianAnd there's nothing you can do about it.
00:31.45JTbrian: you might explode :/
00:31.45Rawplayer:(
00:31.54brianThat's a risk I'm willing to take!
00:32.00JTout there
00:32.04ManxPowe1I'll stick to my Polycoms, thankyouverymuch
00:32.06JTorlok: useful
00:32.16Rawplayerbrian: its a offer
00:32.23EmleyMoorManx: Are you tail-less, or do you have three legs?
00:32.27Rawplayerfrom a dutch grocerystore
00:32.36Rawplayeri've bought that one yesterday
00:32.39Rawplayerit works great
00:32.50Rawplayerso i thought lets share the offer
00:32.54brianmail me one
00:33.00Rawplayer(maybe as christmas present)
00:33.02Rawplayerhehe
00:33.07JT"pricecheck on the Polycom 501 to register 5, pricecheck to register 5!"
00:33.19brianJust switch the tags with a cheaper soft phone.
00:33.21*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
00:33.26ManxPowe1EmleyMoor: it's a play on the Simpson's episode "Max Power"
00:33.40EmleyMoorAh, no wonder I didn't get it
00:33.48brianManxPowe1: hi
00:33.55brianManxPowe1: I would like you to silence my DTMF
00:34.21ManxPowe1*** "brian@*" is now on Ignore List
00:34.32Nuggetinstant silence!  :)
00:34.41*** mode/#asterisk [+o mog] by ChanServ
00:35.19brianWhat did I do :(
00:35.47*** join/#asterisk diclophis-work (n=jbardin@3.170.33.65.cfl.res.rr.com)
00:36.08*** join/#asterisk maverickbna (n=sentinel@wikipedia/Shadowhntr)
00:36.33EmleyMoorIs there a good way I can turn my Caller ID setter into a macro or subroutine in the dialplan?
00:39.54*** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.mn.comcast.net)
00:43.10*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
00:46.41hmmhesays[TK]D-Fender you in here?
00:49.23Dovid~seen [TK]D-Fender
00:49.31jbot[tk]d-fender is currently on #asterisk (2h 56m 7s). Has said a total of 11 messages. Is idling for 2h 43m 15s, last said: 'blitzrage : z0mg!'.
00:49.44macTijn~seen macTijn
00:49.46jbotmactijn is currently on #asterisk. Has said a total of 1 messages. Is idling for 2s, last said: '~seen macTijn'.
00:49.56macTijnoh, right
00:50.22Dovidlol
00:52.47*** part/#asterisk zavala (n=zavala@c-69-180-196-231.hsd1.tn.comcast.net)
00:53.30*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)
00:54.24tuck3ris the --preifx broken in 1.4's configure script?
00:55.16*** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net)
00:55.18EmleyMoorHmmm... something seems badly broken
00:56.31JTwindows style
00:56.46EmleyMoorI tried to call a number and the * box just froze
00:56.57robin__szsure?
00:56.59JTthe whole box
00:57.01JTor just ast
00:57.08EmleyMoorI am sure there's an easy way to make my caller ID setter some kind of subroutine
00:57.11robin__sznormally, you can just ssh in again
00:57.21EmleyMoorSome kind of "virtually everything"
00:57.27EmleyMoorssh seemed unresponsive
00:57.55robin__szhow odd
00:59.49*** join/#asterisk florz (n=florz@2002:58c6:2592:1:0:0:0:2)
00:59.55ManxPowe1sounds like a dialplan loop to me
01:00.10robin__szI had a linux box do that to me once, so I guess it can happen ... that was an early redhat 5.0 box I think, in about 1998/99
01:00.44EmleyMoorManxPowe1: I consider that odds-on!
01:01.17ManxPowe1EmleyMoor: I've been using asterisk for a long time
01:02.09EmleyMoorIndeed, found it
01:04.02*** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-a9020c053d6d7a61)
01:08.00EmleyMoorHowever, for now, I'm off to bed
01:08.08*** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)
01:08.31brianDoes teliax send inband DTMF?
01:08.34*** join/#asterisk predder (n=predder@203.220.55.70)
01:08.45*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:15.56brianI'm using IAX and DTMF seems to still be inband.
01:16.05brianHow do I fix this?
01:16.52[hC]So, i have asterisk set up to allow for blindxfer and attended xfer in features.conf ... What I dont know, is where to specify the context that is to be used when you dial a number to transfer to?
01:17.16brianAgh.
01:17.20brianHelp
01:17.42*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:17.45[hC]Ahh never mind. FOund it.
01:23.30*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
01:24.51brianhi DTMF hates me
01:24.55*** join/#asterisk matt__ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d)
01:25.23*** join/#asterisk ucfMethod (i=ucfMetho@c2.efb7d1.client.atlantech.net)
01:25.28ucfMethodevening...
01:26.06brianI'm getting inband DTMF with IAX.
01:26.22brianHow is this possible?
01:27.21brianI also get inband DTMF with SIP INFO.
01:33.04*** join/#asterisk predder (n=predder@203.220.55.70)
01:34.07jtexter3anybody know if there is an easy way to merge changes from one version to another?  I've made a handful of changes to 1.2.10, and want to upgrade to 1.2.13, but can't get my changes to apply cleanly using a patch file
01:37.29*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
01:37.46brianjtexter3: Manually reapply the patches/
01:38.07jtexter3brian: suck!  I was hoping to avoid that ;-)
01:39.02matt__hiya
01:39.14matt__i am running asterisk on quite a slow pc
01:39.30matt__and my calls take quite a long time to get connected
01:39.50matt__but i use to run asterisk on a linksys router so i know it can run fine on limited memory
01:40.15*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)
01:40.23matt__whats the best way to make it run quicker ?
01:44.43jtexter3brian: is there any easy way to examine the .rej files that get created to determine which components need to be merged manually?
01:45.17*** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com)
01:47.35*** join/#asterisk AvoidingDeadlock (n=ASSERTKI@ppp-70-128-110-113.dsl.tulsok.swbell.net)
01:58.56*** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-69-173-87-106.albyny.adelphia.net)
02:03.01blitzragejtexter3: with 'less'?
02:03.32*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
02:09.25*** join/#asterisk infernix (i=nix@spirit.infernix.net)
02:11.06*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
02:14.20*** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au)
02:16.37*** join/#asterisk lepsie (i=someone@c-68-53-17-135.hsd1.tn.comcast.net)
02:16.44*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
02:19.07*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
02:19.38*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
02:29.55*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
02:40.09*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
02:52.03*** join/#asterisk bkruse_home (n=root@69.73.127.92)
02:54.06*** join/#asterisk dmd1222 (n=dmd1222@c-24-20-35-49.hsd1.mn.comcast.net)
02:55.30matt_my asterix box seems to take a long time to contact remote addresses
02:55.43*** join/#asterisk AvoidingDeadlock (n=ASSERTKI@ppp-70-128-110-113.dsl.tulsok.swbell.net)
02:55.46matt_it prints .. - Executing [107@default:1] Dial("SIP/papport1-086f0000", "SIP/613@fwd.pulver.com|30|r") in new stack
02:55.56matt_and then hangs for about 5 to 10 seconds
02:56.00bkruse_homethen its a network problem.........
02:56.05bkruse_homeits getting a response from fwd.pulver.com
02:56.06matt_and continues
02:56.25matt_its not just fwd tho its all services
02:56.29matt_well it seems to be
02:56.59matt_its defently a network thing cus if i add an exten that answers, plays something and hangs up its nearly instant
02:57.36JTno local hardware extensions?
02:57.51bkruse_homematt_ do a traceroute to fwd.pulver.com
02:58.03matt_humm if i ping a hostname its taking about 5 seconds to resolv
02:58.03bkruse_homejust to satisfy my curiousity
02:58.11bkruse_homematt_ dns server?
02:58.14matt_yea
02:58.20matt_lol
02:58.24bkruse_homedo you have your own dns serveR?
02:58.39bkruse_homeor do you have a lame like linksys router doing 40 different things through switches
02:58.40matt_yea, i have been using the one that comes with openwrt
02:58.41*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
02:58.44matt_dnsmasq i think
02:58.55matt_lol its been good to me so far
02:59.06JTsounds slow as hell
02:59.12bkruse_homematt_ thats disgusting...........
02:59.28bkruse_homeopenwrt has many services running that lil embedded processor cant run
02:59.38matt_humm
02:59.46matt_this pc resolvs instantlly
02:59.53matt_its only the asterisk box that takes like 5 seconds
03:00.06matt_bkruse_home, yea i have taken most of the stuff out
03:00.28matt_ohhhh
03:00.44matt_in the resolv file on the asterisk box i have a nameserver that dosn't exist in the top line
03:00.46bkruse_homematt_ oh what?
03:00.54matt_/etc/asterisk $ cat /etc/resolv.conf
03:00.54matt_nameserver 192.168.7.1
03:00.54matt_nameserver 192.168.4.1
03:01.05matt_... no idea how that got there
03:01.06matt_lol
03:01.20bkruse_home............
03:01.21JTthat often can slow name reolution down
03:01.31bkruse_homeya because it has to try those hosts first.
03:01.40matt_yes
03:01.45matt_its instant .. ish now :)
03:01.55matt_and there i was with strace n everything
03:02.06JTand you didn't even do a ping, gw :P
03:02.29bkruse_home;]
03:02.41matt_no because the latancy when the connection was up was fine
03:02.47matt_it was just the initial connecting
03:02.54matt_so i didn't think to ping
03:03.27matt_is it possiable to get the color terminal when you use asterisk -rv
03:03.27matt_?
03:04.22bkruse_homegood question
03:06.06matt_humm now nothing is connecting
03:06.06matt_ok it is but laggy
03:06.06matt_is there any way to show current connections ?
03:06.26matt_ok i have 2 calls going
03:06.32matt_and the phone isn't off the hook
03:06.52bkruse_homeshow channels.
03:06.52matt_how do i hangup lines ?
03:07.02bkruse_homematt_ http://voip-info.org
03:07.11JTsoft hangup
03:07.48matt_ok
03:08.28matt_i cant see the full channel name
03:08.36matt_it goes off the column
03:09.21Op3rdoes CCNA cert helps in regards on Asterisk??
03:09.40bkruse_homeOp3r: a great voip network is built off a great networ
03:09.45bkruse_homei believe it does
03:09.53*** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
03:09.55Op3rhmmm ok, because Im planning to get one
03:10.20Op3rbut here in my country I cant just go take the exam :( I need to go through CCNA 1234 :(
03:10.21bkruse_homeOp3r: how much do you know now?
03:10.26JTOp3r: good networking knowledge is more useful than a CCNA
03:10.40JTa CCNA does not necessarily give you good networking knowledge
03:10.41bkruse_homejt agreed
03:11.27JTotherwise doing a telecommunications course could help a bit
03:11.42bkruse_homejt: agreed again ;]
03:12.00Op3ri designed and implemented a 300 seater call center (network, station, servers)
03:12.11Op3rso I think im qualified
03:12.13Op3rbut i dont know
03:12.33Op3ri tried the simulation exam and it seems easy though
03:12.34bkruse_homejust because you can plug in network cables and setup basic routing tables doesnt mean your qualified
03:12.48bkruse_homeim not trying to put you down, im just saying never assume you know everything ;]
03:13.08bkruse_homes/put you down/discourage you
03:13.18Op3ryeah I know
03:13.19Op3raheheh
03:13.26bkruse_homek good ;]
03:13.27*** join/#asterisk IronHelix\AFK (n=irc@ool-45785cfe.dyn.optonline.net)
03:13.46Op3rI mean i can do vlans and configure cisco routers with the help of howtos but they keep on demanding i should get a cert
03:13.51*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
03:13.59IronHelixsup all
03:14.19bkruse_homeOp3r: then go for it, tell them to pay to ;]
03:14.21bkruse_homeIronHelix: sup man
03:14.24Op3rbut thats about it
03:14.25IronHelixnot much
03:14.52Op3ri still need to get the ccna 1234 modules
03:14.57Op3rand its going to take a year!
03:15.12IronHelixgot a funky question in case anyone wants to take a shot-  I'm installing * on a shared hosting box.  I compiled / installed Zaptel as root, and I want to install Asterisk as a user.  What files do I have to copy from the root FS to the users jailed FS to get Asterisk to compile w/ zaptel?
03:15.39IronHelixmainly for timing
03:19.16*** join/#asterisk sloth (n=josh@cpe-69-203-212-182.nyc.res.rr.com)
03:21.09*** join/#asterisk leppie (i=leprecha@c-68-53-17-135.hsd1.tn.comcast.net)
03:28.38*** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)
03:31.02*** join/#asterisk rkeels (n=rkeels@c-67-168-55-223.hsd1.wa.comcast.net)
03:31.35FarrisGApparently TelaSip is very good at handling fax over sip trunks. Wish I could figure out exactly what they do that bandwidth.com doesn't so I could possibly convince bandwidth.com to support it.
03:31.56FarrisGThe way I understand it, they really don't need to explicitly "support" fax, there's just something to do with the way the signal is passed along that does or doesn't allow nvfaxdetect to know there's a fax machine on the other end
03:32.16FarrisGFriend of mine is using *, with TelaSip trunks and fax is working beautifully
03:33.46*** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.6.41.Dial1.SanJose1.Level3.net)
03:34.23JTare you sure they're no T38 FoIP?
03:35.11rkeelsthis a good place for noob assistance?
03:35.18IronHelixit can be
03:35.20IronHelixat times
03:35.25IronHelix:)
03:35.29IronHelixwhat can i help you with?
03:35.31bkruse_homerkeels: im in a good mood
03:35.32bkruse_homeso y es
03:35.37aydiosmiohow may we flame you?
03:35.39robl^it depends on how many beers we had
03:35.41IronHelixalso:  did you bring money or food?
03:35.58IronHelixbringing food or especially money can make this place quite helpful... :)
03:36.02rkeelskew: I am trying to get my # box to talk to my sipx box so I can migrate my users off of it
03:36.11JThash box?
03:36.11bkruse_homeaydiosmio: hello, been along time
03:36.26rkeelsAstrix and Trix
03:36.27IronHelixhash box?  i thought that was illegal in most states...
03:36.33FarrisGJT: No, they aren't using T.38
03:36.40bkruse_homerkeels: http://asterisknow.org
03:36.40IronHelixrkeels- you'd probably want to set up a SIP trunk between the two
03:36.45rkeelsNot in Nev I heard
03:36.46JTrkeels: *, asterisk, trixbox
03:36.58IronHelixthen have each one be aware of which extens are on which box
03:37.04rkeelsI tried setting up a trunk but I am not sure I am doing it right
03:37.08IronHelixand if the exten dialed is on the other one, send it there
03:37.23bkruse_homerkeels: http://voip-info.org  is a good start once you get a basic idea
03:37.26JTFarrisG: sounds pretty crazy, it should be hit and miss
03:37.44rkeelsI keep getting a circuits all busy error
03:38.03rkeelshas any one tried doing this before
03:38.08*** join/#asterisk leoncamel (n=leoncame@219.238.107.107)
03:38.16aydiosmiodon't disturb the circuit while they're getting busy
03:38.32bkruse_homeaydiosmio: remember me? kruz?
03:38.43aydiosmiohow could I forget?
03:39.01rkeelsSo all that is necessary is sip.conf with a trunk defined and extensions.conf for dial plan routing right
03:39.12bkruse_homeaydiosmio: good ;]
03:39.28JTrkeels: trunk is a freepbx/trixbox concept
03:39.37JTin sip.conf there are users, peers and friends
03:40.24rkeelsk so in an * box I just need to define peers to talk to sipx then?
03:40.26*** join/#asterisk AvoidingDeadlock (n=ASSERTKI@ppp-70-128-110-113.dsl.tulsok.swbell.net)
03:42.21aydiosmioyou'll never be rid of freepbx.
03:42.53*** join/#asterisk brut- (n=brut@66.173.4.254)
03:43.27ManxPowe1rkeels: #freepbx is the place to ask about freepbx / trixbox
03:44.16rkeelsGot ya....But my Q also aplies to asterisk...I ran into the same promblem with both
03:44.42rkeelsIn * I configured my extensions.conf and sip.conf just like all the docs say
03:44.49rkeelsyet I do not have success
03:45.26rkeelsMy question is; is that all that I need to be configuring to send sip to sip calls to a different pbx
03:45.36ManxPowe1rkeels: put the information I request on pastebin.ca
03:46.10ManxPowe1rkeels: the problem is that none of the standard ways to debug problems work with trixbox/freepbx
03:46.22ManxPowe1their dialplan is SO massivly complex
03:46.56ManxPowe1rkeels: I will only help with the standard asterisk config.  Past JUSt the Dial line to the channel
03:46.59IronHelixagreed, i hate dealing with trixbox.  it's like a pre-osx mac- everything is happy and easy until you need to peek under the hood and fix something.  then god help you.
03:47.30ManxPowe1It should be something like Dial(SIP/${EXTEN}@sipconfentry)
03:47.47ManxPowe1the next priority should be a Noop(HANGUPCAUSE is ${HANGUPCAUSE})
03:47.51IronHelix\AFKThanks
03:48.04rkeelsthat is exactly what I have in my extensions.conf file and I keep getting a dns error
03:48.14rkeelsyet all my dns is set up precisely
03:48.19ManxPowe1rkeels: I'm still waiting for the ACTUAL Dial line.
03:48.21rkeelsas it should be
03:48.25ManxPowe1rkeels: we will get to the DNS error
03:48.29rkeelsall of my digs resolve
03:49.26ManxPowe1First we need to determine if it is a FALSE error.  i.e. indicates it is a DNS error where it might really be a totally different problem that is causing the DNS error
03:49.57rkeelsexten => 2218,1,Dial(2218@sea-na-pbx3251.na.eedinc.net,30,t)
03:49.58ManxPowe1Now, paste the Dial line from your extensions.conf
03:50.24ManxPowe1rkeels: is sea-na-pbx3251.na.eedinc.net a hostname or a sip.conf [sea-na-pbx3251.na.eedinc.net]
03:50.37rkeelsI mean.............exten => 2218,1,Dial(SIP/2218@sea-na-pbx3251.na.eedinc.net,30,t)
03:50.58rkeelsit is host and domain
03:51.15ManxPowe1rkeels: you need to copy and paste or we will waste massive amounts of time on trvial typo or transcription errors
03:51.16rkeelshsot is sea-na-pbx3251
03:51.39*** part/#asterisk FarrisG (n=lckirk@gateway.wiquest.com)
03:51.52ManxPowe1you have a DNS problem.  That host does not resolve for me
03:52.18ManxPowe1rkeels: you do understand that dialing by ip or hostname pretty much bypasses much of sip.conf, right?
03:52.22rkeelsI need to vpn in to get access and then I can't even past it because this comp is on a different network then the station I use to vpn in
03:52.33ManxPowe1rkeels: then I cannot help you
03:52.50*** join/#asterisk bmg505 (n=leon@c1-87-2.rndf.isadsl.co.za)
03:53.00rkeelsThat host is on a private corp network so it wont resolve for you
03:53.49rkeelsgive me ten then and I will c if I can get the exact line.
03:54.25rkeelsNo I am not aware that sip.conf isn't used in this case
03:54.45ManxPowe1Paste the ACTUAL error message off the CLI too.
03:55.32rkeelsk give me five
04:02.12aydiosmioFIVE ISOVER!
04:12.43*** part/#asterisk wiseoldowl (n=Jack@24-236-221-158.dhcp.aldl.mi.charter.com)
04:18.31*** join/#asterisk zamsler (i=zamsler@12.161.149.30)
04:19.55*** join/#asterisk gandhijee (n=akp@static-66-16-235-31.dsl.cavtel.net)
04:20.21*** part/#asterisk gandhijee (n=akp@static-66-16-235-31.dsl.cavtel.net)
04:24.47*** join/#asterisk zamsler (i=zamsler@12.161.149.30)
04:26.38*** part/#asterisk bkruse_home (n=root@69.73.127.92)
04:29.14*** join/#asterisk AvoidingDeadlock (n=ASSERTKI@ppp-70-128-110-113.dsl.tulsok.swbell.net)
04:38.32*** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.mn.comcast.net)
04:41.23*** join/#asterisk rkeels (n=rkeels@c-67-168-55-223.hsd1.wa.comcast.net)
04:43.55rkeelsback
04:44.01rkeelshad power outage
04:48.41*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
04:49.40*** join/#asterisk frk2 (n=fkhan@202.5.145.13)
04:49.49frk2Hi dudes
04:50.01frk2Simple question!
04:50.10frk2how do i do a loopback test on my te110p?
04:50.35*** join/#asterisk Dav1 (n=Dave@71.80.238.59)
04:50.37*** join/#asterisk topping_ (n=topping@207.47.6.130.static.nextweb.net)
04:51.30Dav1wow...  quite the crowd in here...  pretty quiet, though :)
04:51.46frk2hahah
04:51.48frk2anybody?
04:51.51frk2please please
04:51.58JTfrk2: make a loopback cable i'd guess
04:52.04frk2i got the loopback cable
04:52.07frk2its pretty simple
04:52.08DaveKillion+
04:52.09frk2but now what
04:52.15JTzttool i think
04:52.16DaveKillionLoop it?
04:53.08frk2awrite
04:53.21frk2test = controlled inputs  + expected results
04:53.42frk2i know neither the input (maybe zttool) and i definitely dont know what to expect to pass or fail the test
04:54.09JThave you opened zttool yet?
04:54.27frk2oh crack. thats the GUI program?
04:54.50JTit's ncurses or similar
04:54.56JTit does not need x windows
04:55.05JThave you tried to open it yet?  just try it
04:55.07frk2no i didnt spend more than 2 minutes trying to get it to compile
04:55.08JTthat is my advice
04:55.11frk2needs libnewt
04:55.15frk2which i couldn't find
04:55.17JTthen install it
04:55.31DaveKillionJust got my TDM11B last night, and have been playing with it with TrixBox...  fun stuff...
04:55.39frk2im stupid and there was a time we were novell partners, so i installed SLES 9 on it
04:55.59frk2now getting libnewt is a big pain
04:56.06frk2debian is the word on the street
04:56.28DaveKillionGentoo's really good for CLI-based servers
04:56.42DaveKillionbut I've not tried Asterisk on it
04:57.03DaveKillionI whimped out and went 'Box instead
04:58.24JTdebian is god for servers
04:58.28JTgood
04:58.47JTfrk2: i don't see how hard it could be to install libnewt
04:59.07DaveKillionpretty sure I've messed up my config for the analog trunk outbound, however, as I don't get audio out, and I get a timeout from the phone company when I try to dial out
04:59.29DaveKillionI've seen people comment on that before on the boards, but now I can't find those posts
05:02.21*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
05:04.49*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
05:06.02*** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net)
05:06.19DaveKillionI'm doing a phone -> FXO -> Asterisk -> FXS -> Telco ----> Cell Phone test
05:06.47JTwell that's wrong
05:06.56JTphones use FXS interfaces
05:07.02JTtelco uses FXO interfaces
05:07.13DaveKillionsorry, new to the terms, I have it wired correctly
05:07.16DaveKillionI get dial tone
05:07.31JTand FXS interfaces use FXO signalling in asterisk, FXO vice-versa
05:07.33JTah ok
05:07.55DaveKillionI can call from the telset to a SIP phone logged on to the Asterisk server
05:08.13frk2JT not that hard
05:08.15DaveKillionand calls into the trunk from the telco work
05:08.20frk2i just need to pop in CDs
05:08.23frk2which is gay
05:08.35frk2guess i could use YUM
05:08.42*** join/#asterisk TechCentric-Will (n=will@c-71-194-70-13.hsd1.il.comcast.net)
05:08.42JTi see, you could also download the relevant stuff
05:08.43DaveKillion"real men compile" frk2?  :)
05:08.47JTor compile them
05:09.04frk2dude if you can find me the source for libnewt i would be a happy man
05:09.07frk2i have been unable to
05:09.25TechCentric-Willcan someone help me get my cisco 7960 phone working? i just bought it off ebay, i have a tftp server running and my friend gave me all his firmware...trying to get it to talk sip instead of cisco call manager
05:11.27*** join/#asterisk Sephen (n=Sephen@c-69-245-182-37.hsd1.in.comcast.net)
05:11.33DaveKillionoh, and the Digium/Asterisk forum's been compromised - see all the porn spam we've been getting?
05:12.43SephenWhen I call from one Asterisk system to another via IAX2, the Callerid(number) gets transferred, but not the name. Is there some setting I'm missing, or does IAX2 not transfer callerID name?
05:14.04*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
05:22.01QwellDaveKillion: You wouldn't happen to live in MN, would you?
05:23.14DaveKillionSorry, no... Silicon Valley here
05:23.17Qwellk
05:23.26DaveKillionMy mom lives up in TRF, through
05:23.47DaveKilliondifferent last name, though
05:24.40*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)
05:31.43Snake-EyesTechCentric-Will, if i recall correctly, you need some sort of sip lic. from cisco to run if with other systems.
05:32.46TechCentric-Willyou just need the licence to get the firmware, i jave the firmware from my buddy, but i cant seem to get it to push onto the phone
05:33.20*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
05:35.09*** join/#asterisk damien__ (n=damien@gw-morphett.koalatelecom.com.au)
05:35.21SephenTechCentric-Will: We started out with a Cisco 7960 for our Asterisk testing, over 2 years ago. It was a pain to get the firmware and the phone configured correctly. Keep in mind, using that firmware is illegal - the Phones cost around $300 per phone, and then another $140 or some crap for the license for each phone.
05:35.33*** part/#asterisk damien__ (n=damien@gw-morphett.koalatelecom.com.au)
05:35.38*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
05:35.41SephenSwitching to Polycom was the best thing we did.
05:35.43*** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)
05:36.10Strom_CSephen: where the hell were you getting the license from?  last I checked it's like $8 a phone for the license
05:36.33Supaplexstill, they are tEh suck
05:36.40TechCentric-Willim not too worried about the legality of it, i got the software, from a firend, he has the licences, its not totally legal but atleast i didnt otrrent it or anything
05:36.51SephenStrom_C: For the SIP image? Skinny was cheap, I remember that, but not SIP.
05:36.57SupaplexI'd buy em just for target practice
05:37.06Strom_Ci use both cisco and polycom phones, and I'm happy with both of them...although the polycom phones sound muddier than the ciscos
05:37.16SephenIts been close to 2 years since I've looked at the Cisco phones. We've become a Polycom dealer since then.
05:37.41TechCentric-Willi like the polycom and snom phones also, but thats not what i have
05:37.56TechCentric-Willi had a polycom at one point by i dropped it in a move and the screen broke
05:38.06SephenStrom_C: Which Polycoms? I wasn't too impressed with the cheaper Polycoms, but we've had really good luck with the 600s.
05:38.07TechCentric-Willdecided to get something new to play with
05:38.24Strom_CSephen: I've got an IP430
05:38.28TechCentric-Willstrom would you be able to help me get this thing going?
05:38.40Strom_CTechCentric-Will: what is it, exactly?
05:38.51SephenStrom_C: Yeah, thats basically their 300 with a speakerphone in it. We've mainly used the 600s everywhere.
05:39.12TechCentric-Willits a 7960 with firmware version 6.0(4.0)
05:39.15Strom_CSephen: at some point I will have to try the 601 or 650
05:39.31TechCentric-Willneed to get it talking sip since i dont have a callmanager system
05:39.42Strom_Cgot a sip image?
05:39.45SephenStrom_C: I'd like to get a 650 for my office - I hate florescent lighting, so its usally dark in my office. backlit screens are nice.
05:39.49TechCentric-Willyeah all of them
05:40.13*** part/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)
05:40.19*** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)
05:40.47*** part/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)
05:40.48TechCentric-Willi have a tftp server running on my machine, and the firmware in the tftp root dir, i have the tftp server set to my machine in the phone, and its set to use alternate tftp server
05:40.59TechCentric-Willphone boots, and just sits at "configuring ip"
05:41.09TechCentric-Willit has an ip from my dhcp on my linksys router
05:41.33Strom_Cdoes it try and talk to your dhcp server?
05:41.34Strom_Cer
05:41.37Strom_Ctftp server
05:41.43TechCentric-Willas far as i can tell, no
05:41.56*** join/#asterisk juice (n=juice@mo-76-0-46-199.dhcp.embarqhsd.net)
05:42.39Strom_Codd
05:42.42Strom_Cnothing shows up in syslog?
05:43.01TechCentric-Willits actually a windows machine runnig my tftpd
05:43.06*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
05:43.16TechCentric-Willwas told by the same friend that gave me the firmware that this solarwinds tftp server works ok
05:43.16Strom_CTechCentric-Will: you must like pain
05:43.26SephenTechCentric-Will: You have to have a tftp server setup for the Cisco, and that is an option you configure in your DHCP server. Does the Linksys router have the option to specify a tftp server option?
05:43.44TechCentric-Willi doubt it, but i can look
05:44.02SephenTechCentric-Will: If you didn't set it, its not going to know how to get back to it. :)
05:44.19SephenI can paste you my config for our old 7960 if you'd like.
05:44.21Strom_CSephen: he's set the tftp server address manually in the telephone
05:45.24*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
05:45.49TechCentric-Willyes
05:46.56TechCentric-Willwhat should the operation vlan id and admin vlan id be set to?
05:47.18Strom_Ci dont think you need to worry about those
05:47.39TechCentric-Willok just looking at all the settings trying to see if theres something i missed]
05:48.08SephenI sent you a private msg with the configs, did you get them?
05:48.20TechCentric-Willyeah thanks
05:48.27SephenNo prob.
05:48.53SephenThats about all of the detail I can provide on that subject. Like I said, its been 2 years since I've looked at the Cisco phones.
05:49.02TechCentric-Willalright
05:49.04TechCentric-Willthanks
05:49.16TechCentric-Willmaybe ill sell this damn thing and buy a polycom
05:49.17SephenWhen I call from one Asterisk system to another via IAX2, the Callerid(number) gets transferred, but not the name. Is there some setting I'm missing, or does IAX2 not transfer callerID name?
05:50.00Strom_CSephen: IAX2 should transfer calling party name
05:50.11Strom_Care you even setting the name field?
05:50.29*** join/#asterisk yxa (n=lonari@58.185.90.101)
05:51.43*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:52.12yxahow does asterisk get its DIALSTATUS?
05:52.29Strom_Cyxa: it's set by the Dial() application upon exit
05:52.53SephenStorm_C: Maybe thats what I'm confused on. A call comes in via our PRI, gets answered by Asterisk, they hit an extension, which routes the call to another Asterisk box via IAX2 trunking. On the remote Asterisk box, the system shows number only, but not name. On the original box, it shows both. I'm not setting anything, but I'm confused on how to set it, since I'd basically be saying callerID(name) = $callerID(name).
05:53.33Strom_CSephen: do me a favor and pastebin up the relevant iax.conf entries on both boxes and the relevant dialplan logic on both boxes
05:53.40yxaStrom_C it doesn't get it from the opposite party? I mean how does it know when 10 mins of silence is ANSWER or CANCEL etc
05:53.53Strom_Cyxa: answer supervision
05:55.02yxaStrom_C so if i'm using sip, is answer supervision under chan_sip?
05:55.55Strom_Cyxa: I don't think I quite understand what you're asking
05:57.02yxaStrom_C sorry. what i'm trying to understand is how is the DIALSTATUS assigned
05:57.45Strom_Cyxa: the Dial() application sets ${DIALSTATUS} based on progress and supervision messages it receives from the relevant channel driver
05:58.36yxaStrom_C is the sip implementation weak in that? i am getting discrepancies from the bill i received from my ISP
05:58.52Strom_Cwhat kinds of discrepancies?
05:59.40yxabilling discrepancies. minutes log by them an not by us and vice versa
05:59.59Strom_Cwell, obviously
06:00.03Strom_Cbut give me an example
06:00.46yxafor eg, 1 call was logged by them as 50 mins but we only clocked 20
06:01.01*** join/#asterisk stephane (n=stephane@gw.sortilege.net)
06:01.13Strom_Cyxa: are the start times identical?
06:01.46yxamostly.
06:01.53Strom_Cwhat do you mean "mostly"?
06:02.46yxathere are calls which we did not capture at all
06:03.36Strom_Care you completely certain that your PBX is the only device placing calls on this account?
06:03.56yxayeah
06:05.40yxaanother example. we were charged for silence as well. so how does asterisk tell if is answered?
06:06.07JTon pri, easy, Q.931 messages over the D channel
06:06.10Strom_Cit honestly sounds like your telco is on crack.
06:06.26JTon analogue, unless you have answer supervision, asterisk assumes it's answered immediately, for an FXO line
06:07.22yxawe are using PRI
06:07.22*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
06:08.05JTyxa: there are Q.931 messages from the telco to advise a call has been answered
06:08.16Strom_Cyxa: I thought you said you were using SIP
06:08.27yxaits a calling card system
06:08.27Strom_Cyxa: who is your telephone company
06:08.42yxapri for the users, sip for idd
06:09.13yxaStrom_C singtel
06:10.08*** join/#asterisk dlynes_laptop (n=dlynes@s64-180-109-134.bc.hsia.telus.net)
06:11.34*** join/#asterisk dlynes_laptop (n=dlynes@s64-180-109-134.bc.hsia.telus.net)
06:13.33yxaeveryone: if someone were to challenge you on the accuracy of asterisk's ${DIALSTATUS}, what would be your stand?
06:13.49JTfrom all reports, it's pretty inaccurate
06:14.01Strom_Cit depends on what you have it hooked to
06:14.18Strom_Cthe dialstatus is only as reliable as the supervision information passed from the telco
06:14.36Strom_Cso if the telco can't tell a supervised call from it's own asshole, then you have quite a big problem there :)
06:15.49*** join/#asterisk ptblank (n=MURDER1@cpe-75-84-41-226.socal.res.rr.com)
06:15.55*** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com)
06:16.15yxaStrom_C does it help to tell you what commercial equipment my sip trunk is hooked to?
06:16.44JTwait, the stuff you are being billED for, is coming from SIP, isn't it
06:16.58yxajt we maintain our own CDRs too
06:17.08JTso q.931 messages on pri only say if a customer is connected
06:17.09yxaand they don't gel
06:17.27JTno whether your SIP provider believes you are connected
06:17.30JTs/no/not/
06:17.31yxayeah
06:17.54Tondhi I am trying to do call forward using * but am nt getting very far.  I get to save the number in the CFIM db, hwoever not sure what procedure i need to write to check and see if the call forward is on and if so call the forwarded extention or proceed with the original extension
06:18.02Tondany tips is highly appreciated
06:18.15JTTond: freepbx/trixbox?
06:18.28TondAsterisk
06:18.35Tondv 1.2
06:18.41JTwhat is the CFIM db?
06:18.59TondSet(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
06:19.36JTuse of GotoIf in the extension would do it
06:19.41JTthat checks the db
06:19.55JTand sends it to the relevant number
06:20.40Tondso how do i check if that original extension (caller id in the above) ahs any enteries in the * DB?
06:21.49yxaStrom_C so ultimately, DIALSTATUS is passed from the telco and not generated by *
06:22.14JTasterisk interprets messages and indications from the telco/line
06:22.21JTand creates a resultant DIALSTATUS
06:22.41*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
06:22.47yxaJT is the algorithm bugfree in 1.2?
06:22.49JTlooking at the source code for the relevant channel driver (eg chan_sip.c) is probably the best way to see exactly how it's determined
06:22.55JTi doubt it
06:23.09JTbugfree is a very hard claim to make anyway
06:23.10filealgorithm?
06:23.12orlokahh
06:23.18orlokread the fucking source, heh
06:23.21yxaJT well, maybe i should ask if its reliable
06:23.30orlokdont ask that either
06:23.31orlok;)
06:23.34JTi'm not sure, i've heard mixed reports
06:23.36fileif you are talking SIP then your progress for a call should be sent out of band using SIP signalling
06:24.31yxafile "should" ?
06:24.50filedoesn't mean that the remote device will do it as it doesn't "have" to
06:24.55JTmeaning your provider should send accurate indications via SIP
06:25.23JTit's good to make sure you have a reasonable RTP timeout too
06:25.35JTas SIP itself doesn't know if RTP has failed
06:26.17yxaJT what's a good value to set it to?
06:26.26JTi dunno
06:26.36JThopefully your SIP provider has disabled silence detection
06:26.40JTotherwise that can cause issues
06:26.55JT(false hangups during silence)
06:27.54yxaJT there are times that when users call its silent but they actually charge us for it
06:28.19Strom_Cyxa: you realize that something can answer and still be silent, right?
06:29.49yxanod
06:35.20*** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk)
06:36.32yxaStrom_C would it help if i try using SER instead?
06:36.51Strom_CI don't see why.
06:37.11Strom_Cyou need to troubleshoot, not try things blindly
06:38.22*** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-6108e492d908c793)
06:40.24Strom_Cyo rudholm
06:40.38rudholmyo Strom_C
06:47.56*** join/#asterisk VibroMax (n=phisto@83.229.70.154)
06:50.05*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
06:55.06*** join/#asterisk Allanss (i=Allans@host-202-163-247-4.dhcp.infocom.ph)
06:55.28Strom_Cnow it's time to play "track the package"
06:55.35*** join/#asterisk oej (n=olle@apollo.webway.se)
06:55.52rudholmwhat'd you buy?
06:56.48*** join/#asterisk apardo (n=apardo@87.217.147.99)
06:56.59Strom_Caastra 480i
06:57.25*** join/#asterisk borg (n=niklesod@ip70-187-199-233.dc.dc.cox.net)
06:57.52Strom_Ca client is doing a big install with these, so I figure I should have one to dink around with
06:57.58rudholmI prefer progressive-scan myself :)
06:58.05Strom_Chahaha
06:58.09rudholmoh yes, any excuse to buy a phone is good in my book! :)
06:58.46*** join/#asterisk LoveBaby-and-Sti (n=Juliet-@88.224.162.153)
06:59.14rudholmthe keypad looks a lot like a nortel meridian type set
06:59.31Strom_Cyeah, aastra acquired the nortel analog phone division a number of years ago
06:59.38rudholmin fact, the whole phone looks very nortel-ish
06:59.42rudholmah
06:59.46rudholmthat would explain that
06:59.58Strom_Cit rings like a nortel, too
07:00.08rudholmand that's not a bad thing
07:00.37Strom_Cyeah, I like the Nortel ring
07:01.48Strom_Cpleasant yet catches your attention
07:02.09rudholmlike certain humans ;-)
07:02.16Strom_Chaha
07:02.52SephenStrom_C: Thanks again. I'm off to bed.
07:03.00Strom_Cyou're we---ok
07:03.50Chris-NBhi
07:04.01Strom_Chello
07:04.03Chris-NBhow can I install asterisk to a special directory?
07:04.05rudholmhi Chris-NB
07:04.16Chris-NBso I can tar it an move to another box?
07:04.31Chris-NBneed to do that for zaptel, libpri, asterisk and wanpipe drivers
07:04.35Chris-NBanyone knows?
07:05.16rudholmgenerally it's a make option
07:05.32rudholmbut your systems will have to be identical for everything to be "portable"
07:05.32Chris-NBhow/where ?
07:06.02Chris-NBare these options?
07:06.03*** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at)
07:08.45rudholmyou probably want to take a look at your Makefile
07:10.30rudholmwhich Linux distribution are you using, btw?
07:12.10*** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com)
07:14.38*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
07:15.42Chris-NBrudholm, a debian
07:17.03rudholmit might be easier to just use a precomipled apt package
07:17.15*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)
07:17.22rudholmor create one yourself, if you need to customize something
07:18.03*** join/#asterisk ptblank (n=MURDER1@cpe-75-84-41-226.socal.res.rr.com)
07:31.43*** join/#asterisk ptblank (n=MURDER1@cpe-75-84-41-226.socal.res.rr.com)
07:43.24*** join/#asterisk [hC-] (n=hardcore@70.68.154.154)
07:48.09*** join/#asterisk Stanley (n=Stanley@203.111.235.48)
07:48.16*** part/#asterisk Stanley (n=Stanley@203.111.235.48)
07:50.51*** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it)
08:13.21*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)
08:14.21*** join/#asterisk apardo_ (n=apardo@87.217.146.159)
08:15.19*** join/#asterisk gripner (n=leif@195.178.169.154)
08:15.22gripnerhey all
08:15.24gripnerquestion
08:16.24gripneri see something called 6100 - Call que when i view under users,Confer,voicemail. but I DONT see  it under menu option call queues
08:16.41gripnerand i need to remove it, its grayed out under all menues wher i can see it
08:17.48gripneranyone have a bright idea?
08:23.35*** join/#asterisk adeeln (i=adeel@c-24-7-133-237.hsd1.ca.comcast.net)
08:25.30adeelnanyone have any tips onto why i am getting error compilition issues in: chan_zap.c: In function 'zt_call':
08:25.39adeeln?
08:28.12adeelni'm pasting the full error message at: http://pastebin.ca/279743
08:29.31*** join/#asterisk remmo (n=chatzill@smack.isp.net.au)
08:30.57*** join/#asterisk oej (n=olle@apollo.webway.se)
08:38.53*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
08:42.23*** join/#asterisk dacleric (n=dacleric@p54822A4F.dip0.t-ipconnect.de)
08:45.52*** join/#asterisk You_Asterisk (n=younssig@194.204.203.177)
08:55.36*** join/#asterisk CleanerX (n=nix@p54A393D3.dip0.t-ipconnect.de)
08:57.21*** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it)
08:58.39*** part/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it)
09:00.24*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:00.25*** join/#asterisk adeeln (i=adeel@c-24-7-133-237.hsd1.ca.comcast.net)
09:02.49*** join/#asterisk A-Tuin (n=a-tuin@213.237.228.225)
09:03.13*** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net)
09:06.24*** join/#asterisk Rahail (n=rahail1@209-19-88-240.detroit.mi.D-Conn.net)
09:06.26Rahails
09:06.32Rahailany one here who make softphone
09:08.13*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
09:11.15*** join/#asterisk |dennis| (n=dummie@200.32.233.82)
09:17.25*** join/#asterisk apardo_ (n=apardo@87.217.147.239)
09:17.50*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
09:18.16*** join/#asterisk s1gny|wrk (n=s1gny@p54915480.dip.t-dialin.net)
09:18.25*** part/#asterisk s1gny|wrk (n=s1gny@p54915480.dip.t-dialin.net)
09:29.46*** join/#asterisk kippi (n=pssedoff@untrust-gct.equinoxit.net)
09:29.47kippihey
09:30.40kippiI have a incoming call, it gets answered and then plays a message, whist the message is play I would like people to press 1 for exten 101 etc, I have coded this in but its not seeing the key press's
09:36.04Rahailkippi i have same problem
09:36.05Rahail:(
09:38.04kippiit was working yesterday
09:38.25*** join/#asterisk remmo (n=chatzill@smack.isp.net.au)
09:57.01*** join/#asterisk Flusher (i=flusher@filer.euroserv.com)
09:58.44*** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron)
10:00.37hackeronhey, I'm having the following problem, I have the following:    sipgate <> asterisk <> phone-local-network -- There is no firewall on the asterisk box and the phone is on the same network as the asterisk box pointing to asterisk as 192.168.0.1. My problem is one way audio - people can hear me but I can't hear them. Any ideas?
10:02.19*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:04.38kippiI have a incoming call, it gets answered and then plays a message, whist the message is play I would like people to press 1 for exten 101 etc, I have coded this in but its not seeing the key press's
10:05.23*** join/#asterisk ptblank (n=MURDER1@cpe-76-173-170-186.socal.res.rr.com)
10:09.43*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
10:12.17*** join/#asterisk Assid (i=assid@221.134.2.237)
10:12.58*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
10:17.55hackeronI think my problem is the attempting native bridge -- the phones are on the local network with IP addresses 192.168.0.8 and 192.168.0.9, asterisk is 192.168.0.1 and 81.86... --- so how do I stop this attempting native bridge and have everything go through asterisk?
10:19.19Assidhackeron: why would you not want a native bridge?
10:19.29Assidi mean that increases your quality output..
10:19.53hackeronAssid: currently people that call can hear me but I can't hear them :)
10:20.04hackeronAssid: so that's a bigger problem than sound quality at this stage
10:20.21Assidcanreinvite=no
10:20.26Assidadd that to your phones context
10:20.43hackerontrying
10:20.47Assidand your outgoing calls' context
10:20.52Assidin sip.conf ofcourse
10:22.06hackeronbeautiful!
10:23.23hackeronok, so that allowed me to have audio both ways - but asterisk and the phones are on the same network, the asterisk box is the gateway so how do I get it to work without canreinvite=no?
10:24.01Assidwhats wron with it now?
10:24.21hackeronnothing, it works with canreinvite=no, but not with canreinvite=yes
10:24.28Assidofcourse not
10:24.40hackeronlol
10:24.54*** join/#asterisk tmyneii (n=localhos@161.53.107.100)
10:24.58Assidcanreinvite lets the phone connect to the outside.. and apparently the location you are calling has a nat too..
10:25.20Assidso the opposite * server gets confused if it receives the nat'd phone ip
10:25.21hackeronit has a nat, but no firewall and I had nat=yes
10:25.43Assiddoesnt matter.. the ip it realizes its on is the nat'd ip
10:25.59Assidmake sure you modify the sip.conf  to mention the 'externalip'
10:26.18hackeronwhat if the externalip is dynamic?
10:26.35Assidthen use a host.. like dyndns
10:26.47Assidexternhost i think is the directive
10:26.52Assidread up the comments in sip.conf
10:26.55hackeronah, excellent
10:27.05Assidbut you should leave canreinvite=no
10:28.13kippiI have a incoming call, it gets answered and then plays a message, whist the message is play I would like people to press 1 for exten 101 etc, I have coded this in but its not seeing the key press's
10:29.25Assidare you using play() or background()
10:33.05hackeronAssid: ok, thanks for the help - the quality is really good even with canreinvite=no
10:33.16Assidyep
10:33.26*** join/#asterisk emiquelito (n=emiqueli@200-155-185-1.static.spo.ifx.net.br)
10:33.30Assidas long as you dont resample.. your fine
10:33.47Assidyou calling from 1 box to another? or to some providers terminator
10:33.51kippiworking!!
10:33.55kippiThanks Assid!!
10:34.20hackeronAssid: calling to some provider/terminator
10:34.37Assidwhich one?
10:34.43hackeronAssid: pipecall and sipgate
10:34.43Assidif you dont mind me asking
10:34.46Assidk
10:35.02Rahailkippi
10:35.06Rahaildid you get it work
10:35.12Rahailwhat you did please tell me
10:35.13hackeronAssid: pipecall is very cheap outgoing, and sipgate is cheaper mobile calls and local incoming
10:35.57Assidhrmm uk based
10:36.05hackeronyep
10:36.32Assidlegend.co.uk rihgt?
10:36.39Assidthats going realllllly slow for me
10:36.41*** join/#asterisk vgster` (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
10:37.10Assidkippi: usng the wrong function ?
10:37.59*** join/#asterisk vgster` (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
10:39.24hackeronAssid: yeah, legend
10:39.31*** join/#asterisk soylentgreen (n=fgast@193.238.89.34)
10:42.08*** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br)
10:42.42*** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr)
10:44.31Chris-NBhi
10:44.37Rahail./
10:44.48Chris-NBanyone played around with presence watcher/ing ?
10:45.09Chris-NBespecialy with grandstream gxp-2000
10:45.35*** join/#asterisk Guest^DJ (n=me@211.24.146.11)
10:45.48Guest^DJhi anyone work with adit600 before?
10:48.28*** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it)
10:55.21adeelnhas anyone compiled asterisk extensions with uclibc ?
11:03.14*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
11:07.32*** join/#asterisk daysmen3 (n=primus@host81-158-206-37.range81-158.btcentralplus.com)
11:09.00*** join/#asterisk rcsw (n=richard@mail.shout-telecoms.com)
11:23.04*** join/#asterisk mega (i=mega@gateway/tor/x-56e62a70ce49ddac)
11:25.59*** join/#asterisk bmg505 (n=leon@c1-87-2.rndf.isadsl.co.za)
11:26.24*** join/#asterisk stephane (n=stephane@gw.sortilege.net)
11:28.40*** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt)
11:28.54*** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com)
11:29.46in-ptHi all
11:29.52*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
11:30.32in-ptI am getting some weird logs on asterisk cli.if i dials on a extension which is offline "482 loop detected"
11:31.09in-ptwhats the reason for that ..and it looks for the extension in local context.but i dont want local context to come in my dial plan
11:31.18in-ptany suggestion..anyone ?
11:33.17*** join/#asterisk zapp-branigan (n=zapp-bra@81-202-145-20.user.ono.com)
11:35.21*** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com)
11:37.27*** join/#asterisk henrique (n=henrique@201-27-71-35.dsl.telesp.net.br)
11:43.17*** join/#asterisk xnon (i=xnon@200.8.5.123)
11:49.52*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
11:50.36*** join/#asterisk bmg505 (n=leon@c1-87-2.rndf.isadsl.co.za)
11:51.46*** join/#asterisk Skarmeth (n=Skarmeth@201009104156.user.veloxzone.com.br)
11:53.04ghenrywhat's the point of AsteriskNOW? When there's trixbox. Why are Digium trying so hard to steal Trixbox contributors and users?
11:55.30*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
11:57.40*** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net)
11:58.10Skarmethhi all
11:59.29kippiis there away to make say 10 extenstions without making them agents?
12:00.51Hermione_why not?
12:01.01Hello2007<Hello2007> i have  a problem that one someone call me and hang up before i answer, asterisk still process the call and send it to the voicemail,and when i open my voice i get a message with dial tone busy
12:01.02Hello2007<Hello2007> in the message i hear the busy dial tone
12:01.17Hermione_101, 102,103,...,110.
12:01.46SkarmethI am trying to get lastest (trunnk) gastman source code from Digium's SVN repo (svn checkout http://svn.digium.com/svn/gastman/trunk gastman) but I only get a "svn: REPORT request failed on '/svn/gastman/!svn/vcc/default' <newline> svn: REPORT of '/svn/gastman/!svn/vcc/default': 400 Bad Request (http://svn.digium.com)" fail message
12:04.58kippiHermione_ Like this? exten => s,3,Dial(sip/1130,1131,1133,1134,1135,1136,1138,1141,1139)
12:05.32*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
12:06.25*** part/#asterisk henrique (n=henrique@201-27-71-35.dsl.telesp.net.br)
12:07.07Hermione_kippi: in this case they will ring all together
12:07.56kippihmm they dnt seem to
12:08.57Hermione_kippi: sip/1130&sip/1131&sip/1133......
12:11.30monstedurgh, windows support is evil
12:11.40monstedmostly because of the users
12:11.42monstedi hate users
12:11.56monstedlife as an admin would be so much easier if we had no users
12:12.46Hermione_monsted: but only servers...
12:14.28Skarmethmonsted, if sys/net admins don't have users, we don't have jobs... who will break things?
12:14.33*** join/#asterisk kashmish_ (n=kashmish@m1.ince.net)
12:15.16monstedSkarmeth: don't get all realistic on me
12:15.39monstedi want a world with no pesky users where i have lots of time to play world of warcraft
12:17.02SkarmethMy dream are watch users walking in a indian queue in a mountain and jumping inside of an active volcano...
12:17.16monstedlemmings!
12:18.21Skarmethdoes someone has a tarball of gastman latest source?
12:18.34SkarmethI can't get it from Digium SVN...
12:27.27kippiHermione_ That config gives me this error Dial argument takes format (technology/[device:]number1)
12:29.43*** join/#asterisk zotz (n=zotz@24.244.163.157)
12:29.50*** part/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
12:30.18kippigot it working :)
12:30.40*** join/#asterisk bmg505 (n=leon@c1-87-2.rndf.isadsl.co.za)
12:36.48kippiif I just want to make the extenstions ring and ring and nerver stop until someone picks up what would I want to put at the end of my dial command
12:40.37*** join/#asterisk eddi (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
12:45.48*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
12:49.47*** join/#asterisk benjamin (n=benjamin@62.80.0.226)
12:49.48*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)
12:49.52benjaminhello
12:50.14benjaminis there anyone who can help me with callforwarding in freepbx?
12:51.25*** join/#asterisk vaineh (n=evaing@82-71-116-46.dsl.in-addr.zen.co.uk)
12:53.31benjaminhello
12:53.40benjaminis there anyone who can help me with callforwarding in freepbx?
12:54.52Nivexbenjamin: there's a #freepbx channel afaik.  They can probably help you better with the specifics of that platform.
12:55.34Nivexhmm... maybe not... must be thinking of another platform
12:55.52*** join/#asterisk enema_cow (n=enema_co@cugnet.net)
12:55.57benjamindo you know wich platform?
12:56.03benjaminok
12:56.06benjaminthanks
12:57.52*** part/#asterisk benjamin (n=benjamin@62.80.0.226)
12:58.45*** join/#asterisk paljas (n=paljas@sarastro.cs.uu.nl)
13:00.00*** join/#asterisk santibiotico (n=santi@101.Red-83-58-114.dynamicIP.rima-tde.net)
13:00.04santibioticohi
13:01.07*** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk)
13:02.04santibioticoi have a little problem with zaptel and quadbri...it all is working fine, but when i dial any number, i can hear two dial tones
13:02.41santibioticoany idea about how to hear only one tone
13:02.49santibiotico:?
13:04.39*** join/#asterisk jerryoc (n=jerryoc@cpe-75-80-102-22.socal.res.rr.com)
13:05.59*** join/#asterisk andresmujica (n=andresmu@201.245.228.228)
13:09.05vainehim using a digium tdm400p on trixbox with nokia e61's on the wireless network acting as sip phones. when i call in externally it rings twice before my sip phones ring.. anyone got any idea why?
13:09.37*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
13:10.41jerryocmaybe the progressinband option me thinks
13:10.48jerryoctried that?
13:12.48*** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net)
13:18.21*** join/#asterisk blablub (n=bla@217.13.167.28)
13:19.31*** part/#asterisk blablub (n=bla@217.13.167.28)
13:20.01*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
13:20.37*** join/#asterisk QbY (n=Kelvin@cm-64-221-168-170.dhcp.southerncoastalcable.net)
13:20.56QbYIs there an Asterisk function which will check for the presence of a channel variable?
13:22.52*** join/#asterisk nortex_work (n=breeves@snapper.titanspecialties.com)
13:23.00Hello2007does anyone one what these field do in zapata.conf?
13:23.03Hello2007prewink: Sets the pre-wink timing.
13:23.04Hello2007preflash: Sets the pre-flash timing.
13:23.10*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)
13:23.52*** join/#asterisk eddi (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
13:23.54Hello2007whts the wink and whtas the flash?
13:33.52*** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk)
13:34.21*** join/#asterisk [Airwolf] (n=airwolf@89.205.155.84)
13:34.28*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
13:35.09*** join/#asterisk navigo (n=navigo@adsl-230-134-184.gnv.bellsouth.net)
13:36.09*** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)
13:36.16navigois there a searchable index of the asterisk mailling list?
13:36.33QbYnavigo:  google.com -> site:lists.digium.com
13:36.33tmccraryDo regular FXO gateways work in China?
13:36.43*** join/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy)
13:36.57navigothanks.
13:37.17punkgodeanyone knows if attended transfers in a queue should be logged in queu_log ? or just blind transfers are supported?
13:38.15*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
13:38.50punkgodeI'm using both kind of transfers, and can't get attended transfers to log in queue_log... like it never happened
13:40.38*** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)
13:45.54navigoanyone here have any experience trying to project callerid down a pri on which the asterisk interface is in pri_net mode?
13:46.13navigowhen the pri is in pri_cpe mode, I can project fine.
13:46.36DrukenHMEi would assume it's the same, set it and send it
13:46.57navigounfortunately not.
13:47.18navigoI have used a couple of atlas 550s to sniff the traffic and the setup of the call is even different.
13:47.48navigowhen the asterisk interface is in pri_cpe, it sends the CID Name in the setup message as a facility message IE-1C
13:47.50*** join/#asterisk javar (n=javar@69.79.134.24)
13:48.36navigowhen the * interface is in pri_net, it sends the CID Name as a Display message, which seems to not be read by most KSU equipment.
13:48.54navigo(that is most as in the equipment I have available to test with here)
13:49.07*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
13:50.18mendolguys can I connect 2 analog lines to SPA9000 and forward them to SIP trunk?
13:52.07*** join/#asterisk adorah (n=admin@84.94.123.173)
13:52.28*** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net)
13:54.03DrukenHMEmendol: from what i see the spa9000 is FXS only
13:56.23mendolso any linksys with fxo support?
13:56.35mendolso i could connect analog line, convert it and send to sip trunk?
13:56.37*** join/#asterisk sloth (n=sloth@pool-162-84-157-242.ny5030.east.verizon.net)
13:56.39DrukenHMEyep....
13:56.55*** join/#asterisk qwertz (n=qwertz@pD9532A13.dip0.t-ipconnect.de)
13:57.49DrukenHMEbut i don't use them....
13:58.03mendolwell its much more complicated
13:58.09mendoland have to solve that problem
13:58.42DrukenHMEuse x100p cards...
13:59.03DrukenHMEor have a voip carrier port them....
13:59.48mendoli need to find cheap solution for connecting 2 departaments of same company
14:00.24mendolone has telephone-exchange and need to connect it to 2nd departament using voip
14:00.36qwertzHi, when I use the "zap show channels" I can see the incoming calls but not the outgoing ones - so is this some kind of misconfiguration on my side or is this intended behavior ( and if yes how can I get a complete list of all zap channels in use atm on a * 1.0.10)?
14:00.38*** join/#asterisk WoLF (n=AnaStaSy@88.240.45.138)
14:00.42tmccraryDo any of you guys know much about China's PSTN and POTS lines there?
14:02.19DrukenHMEisn't china much like canada as for pots?
14:03.09tmccraryI don't know, is it? :(
14:03.35tmccraryMy issue is I am getting one way audio in China with an FXO gateway. I can here the person fine, but the person in China can't hear me
14:03.49tmccraryI'm not going through a NAT and voip-voip calls work fine
14:04.09tmccraryThe issue seems to be at the gateway/phone lines
14:05.23DrukenHMEcould it be the tx level on the device?
14:06.40*** join/#asterisk melllwa_ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
14:06.56tmccraryhmm, let me try, thanks
14:08.01*** join/#asterisk DirtyD (n=DirtyD@ool-44c2dcca.dyn.optonline.net)
14:08.02DirtyDHi.
14:08.14*** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk)
14:08.14DirtyDDoes anyone know if asterisk has any CALEA support?
14:08.25tmccrarywow, one line just gives me an explosion of static
14:08.36tmccraryChina's phone lines are crazy
14:10.36*** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-2fe888b7624d99f2)
14:11.51*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
14:14.56*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
14:15.05tzangerok maybe a dumb question... I've enabled the http server in Asterisk (trunk from yesterday) -- I can hit it on port 8088 but I can't pull up anything at all
14:15.29Katty:<
14:15.45Kattymy boss scheduled me to be in a meeting...with the phone company...cause apparently they have a lil centrex thingy.
14:15.59Kattyi can't believe they're even entertaining this guy.
14:16.01tzangerhttp://ip.of.asterisk.box/asterisk/ajamdemo.html, /asterisk/static-http/, /static-http/, /... can't find it
14:16.24tzangercentrex is well-established and gives the boss a warm fuzzy feeling
14:16.54Kattythis company is not well established...and it gives me a haty churning feeling inside.
14:17.03Kattythey couldn't get voIP working after 2 months of work...
14:17.04tzangerKatty: but centrex is, that's the point
14:17.10Kattyit took them 3 months to get our t1 turned on
14:17.15Kattyi wanna strangle these people.
14:17.45Kattytzanger: can it do anything asterisk can't?
14:17.54tzangeryes and no
14:17.56Kattytzanger: does it run on pretty linux? :<
14:18.09tzangercentrex can get you your rollover and stuff that you cannot get easily with 1FL (normal business) lines
14:18.25tzangerbut if you're pulling your calls in over a DID or are willing to play with centrex's hookflashing, you can make it work
14:18.31tzangeryou don't want to play with centrex's hookflashing, btw
14:18.39Kattyis it skeery?
14:18.41tzangerno it doesn't run linux, it runs on the telco switch
14:18.46Kattyeww
14:18.52Kattybutbut
14:18.53Kattybut
14:18.54*** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com)
14:19.02Kattyhow's it going to talk to the jabber server and spam people )=
14:19.10tzangerit won't
14:19.16Kattyandand, how's it going to ssh over to apollo and turn the music down
14:19.20Kattyand play audio files over the speakers
14:19.25Kattywith pretty public ssh keys :<<<
14:19.54QbYIf I have:exten => s,2,Set(TIMEOUT(digit)=3)
14:19.54QbYexten => s,3,Set(TIMEOUT(response)=7)
14:19.54QbY...and WaitExten(20) -- shouldn't asterisk drop me to the t,1 priority after 7 seconds?
14:20.01*** join/#asterisk AuPix (n=root@mail.aupix.com)
14:20.05Kattytzanger: i think this is all rubbish.
14:20.08Kattytzanger: i want my bash.
14:20.14tzangernah it's just the old way
14:21.13mendolany linksys i can use to convert fxo incoming signal to voip?
14:21.59tzangerpap2na?
14:22.04tzangeror is that fxs
14:22.05tzangerI can't remember
14:22.07tzangerI don't use that stuff
14:23.40*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)
14:24.12mendolaha
14:27.14mendoli need smth which can convert my fxo signal to ip :-/
14:30.06DrukenHMEtzafrir: pap2 is FXS
14:30.45DrukenHMEer.. god damn nick complette.. tzanger!
14:34.38*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:34.38*** mode/#asterisk [+o anthm] by ChanServ
14:36.18DirtyDIs there any CALEA support for Asterisk?
14:43.03*** join/#asterisk sloth_ (n=sloth@64.3.170.41.ptr.us.xo.net)
14:45.49*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:45.57blitzrageDirtyD: you build it yourself
14:45.58vainehwheres the main log file for asterisk/trixbox?
14:46.04blitzragei.e. ChanSpy()
14:46.12blitzrageand MixMonitor()
14:46.28*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
14:46.39blitzragevaineh: ask in #freepbx
14:47.03vainehrgr
14:48.54tmccraryCALEA is stupid :)
14:49.15*** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it)
14:49.46*** join/#asterisk pigpen2 (n=mark@207.71.33.114)
14:50.53*** join/#asterisk |dennis| (n=dummie@200.32.233.82)
14:51.54*** join/#asterisk dasenjo (n=dasenjo@190.24.176.58)
14:52.00*** join/#asterisk sloth (n=sloth@64.3.170.41.ptr.us.xo.net)
14:52.20shellsharkDirtyD: from the way i understand it from www.askcalea.net, as long as your system is able to unobtrusively record a call when law enforcement requests you to, you are CALEA-compliant
14:52.35shellsharkhttp://www.askcalea.net/capability.html
14:53.16shellsharkthere are 4 main points there
14:53.16shellsharki think asterisk covers all of them out of the box
14:53.17ChkDigitSo ChanSpy and Monitor make asterisk CALEA compliant?
14:53.17coppicewhich is impossible in a VoIP environment
14:53.27ChkDigitUnless you keep the server in the media path.
14:53.41ChkDigitAnd your users don't know how to get around it.
14:53.50coppicewhich would be really stupid
14:54.14shellsharkwhich would be really smart, if you want to be calea-compliant ;)
14:54.40coppicewhich gets back to "CALEA is stupid"
14:54.47shellsharkChkDigit: for an ITSP with re-invites off, it's very simple
14:54.50ChkDigitIt is just on request by law enforcement, right?
14:54.55shellsharkChkDigit: right
14:55.09shellsharkcoppice: no doubt about that, but we've got to abide by it
14:55.15ChkDigitSo having a few number of devices forced to stay in the media path is not a big deal.
14:55.40shellsharka simple AGI should suffice
14:55.52ChkDigitThe law if probably not going to say, "I want every call from everyone in your company, all the time."
14:55.58shellsharkcheckifuserisbeingwatchedbybigbrotherandmixmonitoraccordingly.agi ;)
14:56.10coppiceI'm sure vonage's looses would be wondefully affected by all the audio going through their servers :-)
14:56.16ChkDigitYou forgot the macro- in front.
14:56.36shellsharkChkDigit: that was the AGI filename ;)
14:56.45ChkDigitMy bad. =)
14:56.51Kattyfile: today i'm gonna setup a jabber server, me thinks!
14:57.01Kattyfile: if i can figure it out, anyway.
14:57.27shellsharkcoppice: Vonage does a re-invite from the CPE directly to the PRI or what?
14:59.04tzangerfindlay: who's the person you're with in your pic?
14:59.08tzangerer file not findlay
14:59.10Kattyfile: do you have a shiny apt-gettable jabber server?
14:59.12*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
14:59.19KattyZeeek: !
14:59.26fileKatty: nope
14:59.28Zeeek{{{{Katty}}}
14:59.31Kattyfile: awe )=
14:59.52Zeeekjust what I need, several geeks eager to help me with a stupid question
15:00.13*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
15:00.28ZeeekI'm trying to dial a sip URI: dial(SIP/yermutha@domain.tld,30) ;;; for example
15:00.43ZeeekI get no route error
15:01.28Zeeekactually it's more like         dial(SIP/canon.yermutha@domain.tld,30)
15:01.30*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
15:01.36Zeeekoh... crap
15:01.47Zeeekactually it's more like         dial(SIP/yermutha@canon.domain.tld,30)
15:01.57puzzledhi
15:02.08*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)
15:02.24Zeeekthe domain canon.domain.tld does NOT ping but it does work in other SIP phones. Is this a SRV trick?
15:02.31fall0utHrm, I need some cheap SIP termination for like 25-30million mins/month
15:02.33fall0utwho does that?
15:02.44puzzledlevel3
15:03.16*** join/#asterisk juanjoc (n=juanjoc@201.216.212.113)
15:03.35shellsharkfall0ut: for that kind of volume you could get some real cheap rates...
15:03.36blitzragewill L(3) even take someone with that few minutes anymore?
15:03.46shellsharkfall0ut: check with Level3 and XO
15:03.49ZeeekKatty hug appreciated but no answer ?
15:04.03shellsharkblitzrage: they only have 1 million minute / month minimum
15:04.10blitzrageahhhh gotcha
15:04.28shellsharkblitzrage: 25-30 million minutes is nothing to sneeze at ;)
15:04.38blitzrageheh :)
15:04.51fileblitzrage: !
15:04.52Zeeekhi blitzrage by the way
15:04.56blitzragefile: !!!
15:04.58blitzrageZeeek: !!!
15:05.07pfhorgeseriously. That's 694 calls every minute of the month
15:05.17fileAPC is on my hate list
15:05.18blitzragepfft... not even 700 :)
15:06.01pfhorgemore than my office needs, with its 2 mighty PSTN lines
15:06.07shellsharkfile: why is APC on your hate list?
15:06.15blitzragewow... who'd a thunk that creating an alarm clock could be so complicated
15:06.37shellsharkblitzrage: a one-time alarm clock is no big deal
15:06.37filethey woke me up early for a survey
15:06.41blitzrageI figured that was gonna take me like 2-3 hours... instead it was more like 2 days and 100 lines of dialplan logic and 2 PHP scripts
15:06.45shellsharkblitzrage: recurring alarms are a major PITA
15:06.49blitzrageshellshark: this isn't a one-time alarm clock
15:06.52shellsharkfile: ah ;)
15:06.59blitzrageyah -- I built recurring, with snooze, and timezones
15:07.13shellsharkblitzrage: can you hook me up?
15:07.16blitzrageattached to a DB, with failover inside a cluster
15:07.26shellsharkurl me ;)
15:07.30blitzrageshellshark: probably won't make much sense since its part of my own DB :)
15:07.39shellsharkerr, integrated?
15:07.48blitzrageyah.. integrated into a clustered, vPBX platform
15:07.59shellsharkintegrated into the DB i mean
15:08.13shellsharkyou're using SQL standard queries, right?
15:08.31blitzrageyes, information is stored into rows in a PGSQL database, and information read/written from func_odbc directly into the dialplan
15:09.08shellsharkoh i never thought about doing that.... i was trying to use an AGI
15:09.20shellsharki like to keep the dialplan as simple as possible
15:09.32shellsharkfork all the difficult logic stuffs to AGI scripts
15:09.42blitzrageAGI is too much overhead... func_odbc is less
15:09.57blitzrageplus func_odbc has failover DB now
15:10.20blitzrageI do some pretty crazy things in the DP now :)
15:10.30pfhorgeAnyone had problems with phantom ringing / ring debounce in an ATA?
15:11.01blitzrageshellshark: wait for March or April... it might show up in a book :)
15:11.32fileblitzrage: pfft
15:11.43blitzragepffffft
15:11.55blitzragefile: you are soooo fired
15:12.10fileyou are totally not my boss
15:12.18shellsharkfire! fire! fire!
15:12.21blitzragemaybe not... but I'm the boss OF you
15:12.30shellsharko_O
15:12.32blitzragenow go do my bidding!
15:12.38blitzrage(on ebay)
15:12.43*** join/#asterisk Assid (i=assid@221.134.2.90)
15:13.03shellsharkfile: how does your +o react to such vulgarities? ;-)
15:13.16shellsharkbah
15:13.19shellshark;)
15:13.19mercestespfhorge:  Check your voicemail message waiting indicator.  If "phantom rings" when you have a voicemail.  Most common cause.
15:14.11mercestespfhorge:  It's a *feature*.
15:14.45pfhorgemercestes, I saw that in Digium's knowledge base, but we don't have voicemail on the line
15:15.28mercestespfhorge:  Hrm...which ATA?
15:15.39pfhorgeTDM2400
15:15.54*** join/#asterisk |dennis| (n=dummie@200.32.233.82)
15:16.05*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)
15:16.16shellshark... nice ATA ;)
15:16.22mercestes...  Yea.
15:16.33mercestesHow many of those ATA's do you have?
15:16.40shellsharkmercestes: it's not an ATA
15:16.45*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
15:16.48mercestesshellshark:  I know.
15:17.02pfhorgeI thought ATA was analog telephone adapter?
15:17.09flujanhi guys... I am searching for a good softphone that supports dial from browser. Like idefisk-biz.
15:17.23flujanDo you have recomendations?
15:17.33flujanI can also use the SIP protocol.
15:17.52Rawplayeranyone in here running * behind a obsd firewall with nat?
15:17.59shellsharkflujan: nat
15:18.00mercestespfhorge:  It is.  <mercestes> shellshark:  I know.
15:18.00mercestes<pfhorge> I thought ATA was analog telephone ada
15:18.07mercestes...
15:18.07blitzrageflujan: moziax
15:18.08Dr-Linux|work~dict dejuidure
15:18.09mercestesdamnit.
15:18.14mercesteshttp://www.mconnectinc.com/images/Linksys_PaP2.jpg
15:18.16mercestesthere.
15:18.29Dr-Linux|work~dict human
15:18.44*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-246-217-92.washdc.fios.verizon.net)
15:18.48Dr-Linux|workjbot wakeup
15:18.59*** join/#asterisk pifiu (n=someone@216.5.79.1)
15:19.04SomeOne1i'm experiencing a lot of static... how should i trouble shoot that?
15:19.20mercestesSomeOne1:  Are you using SIP or PRI?
15:19.28shellsharkSomeOne1: or BRI or POTS?
15:19.38pifiumorning everyone
15:19.45tmccrary~dict dictionary
15:19.45pfhorgeapparently it's only an ATA if it the device goes directly from PSTN to VOIP?
15:19.46shellsharkpifiu: mornin
15:19.51tmccraryah yes
15:19.54tmccrarynow I remember
15:20.00shellsharktmccrary: lame
15:20.04SomeOne1SIP
15:20.06SomeOne1and POTS
15:20.27shellsharkSomeOne1: does your POTS card have echo cancellation?
15:20.43SomeOne1wait no sorry i thought you were saying something else
15:20.46SomeOne1i dotn have a POTS card
15:21.00SomeOne1my asterisk server is reciving calls from an origination provider
15:21.07SomeOne1and is playing an mp3 for now
15:21.13mercestesSomeOne1:  Via sipconnect or via a T1?
15:21.16SomeOne1and theres like so much static its riduculous
15:21.20SomeOne1a T1
15:21.35shellsharkSomeOne1: so you have a T1 card?
15:22.01SomeOne1heh, its verizon FiOS (fiber) to my house, its pretty decent bandwidth
15:22.08SomeOne12.5mbps upload, 15mbps download
15:22.22shellsharkso you're not using a T1 connect ;)
15:22.26SomeOne1and its a box that just converts it to a cat-5
15:22.31SomeOne1im not
15:22.35SomeOne1heh
15:22.37SomeOne1sorry
15:22.51mercestesSomeOne1:  Then it's likely not your problem.  Complain to your provider.
15:22.55pfhorgeWe have Cat-3 here. No seriously, it's 3 cats. We just tape the bits on.
15:22.58shellsharkSomeOne1: yeah, provider issues
15:23.15*** part/#asterisk [Airwolf] (n=airwolf@89.205.155.84)
15:23.17flujanblitzrage, I am currently using moziax. Any other idea? :)
15:23.18*** join/#asterisk hohum (n=dcorbe@host-12-195-58-235.iad1.interceltelecoms.net)
15:23.20SomeOne1plus i got like
15:23.25mercestesSomeOne1:  But first....you should dig up your fibre and make sure you don't have water on your fibre..because that can cause static causing shorts.
15:23.35blitzrageflujan: nope
15:23.38SomeOne1heh,
15:23.40shellsharkmercestes: rofl
15:23.44flujanshellshark, I never heard of this softphone... :( Where can I grab it?
15:23.44SomeOne1water cant affect fiber
15:23.45SomeOne1like that
15:23.45blitzrageLOL
15:23.52SomeOne1:P
15:23.57tmccraryalso, check your FM theory, you may need more FM
15:24.03flujanblitzrage, thanks. :)
15:24.22shellsharkflujan: eh? i was mainly joking... but jain is a software toolkit that you can develop java-based soft phones with
15:24.26mercestesI always love it when VoIP customers complain about "static on their lines."
15:24.31SomeOne1could it be a, like, codec conversion problem
15:24.56SomeOne1which is degrading the quality or not converting it right so theres like "static"
15:24.58mercestesthen I tell them "you don't have lines...you have a data T1 and SIP."  "Well, on my phone then."  
15:25.10shellsharkmercestes: haha
15:25.12mercestesMa'am, unless you submerged your headset in water, *you* don't have static.  Go away.
15:25.19nays85don't overthink it... just try the obvious: try it with a different handset, try it with a different ATA, try it with a different SIP provider
15:25.30SomeOne1nays85: you talking to me?
15:25.33SomeOne1cool, yeah i guess
15:25.54SomeOne1but im not sure if a codec conversion will be causing that, i guess probably not
15:26.00shellsharkOH NOES! TEH STATIC IS IN MUH FONE!
15:26.29shellsharkhehe
15:26.34blitzragemercestes: I tell them to turn the volume down on their phone because the other person is talking too loud
15:26.54nays85well, if it happens with a different phone and a different SIP provider -- it's your asterisk box or ATA
15:27.18nays85dropouts might be from your ISP but not static
15:27.28SomeOne1yeah no dropouts
15:27.31SomeOne1definately static
15:27.40SomeOne1so like, yeah its probably the SIP origination providers fault
15:28.10shellsharkSomeOne1: nub ;)
15:28.20nays85so try another provider or another phone number... it could ultimately be the originating PSTN switch where that phone number comes in from
15:28.21SomeOne1whats a nub? :(
15:28.30shellsharkSomeOne1: newbie
15:28.33shellsharkheh
15:28.36SomeOne1:'(
15:28.43nays85your third leg is a nub
15:28.48SomeOne1sorry i let you down shellshark
15:29.00shellsharkSomeOne1: lol, everyone's gotta start somewhere :)
15:29.16SomeOne1also, why cant i get logging or verbose console to show like, what codec the calls are using, i thought it would be obvious to put that there
15:29.28shellsharkSomeOne1: sip debug
15:29.34SomeOne1i have messages on full in logger.conf
15:29.34SomeOne1messages => debug,notice,warning,error,verbose,dtmf
15:29.35shellsharkwill show you codec negotiation
15:29.41SomeOne1i see
15:29.52mercestesit's spammty
15:29.59*** join/#asterisk marv[work] (n=timr@24.214.206.254)
15:30.12SomeOne1ahhh, i see
15:30.16SomeOne1sipdebug = yes in sip.conf
15:30.18SomeOne1duh!
15:30.31*** join/#asterisk ming_zy2 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
15:30.37mercestesduh...type it in the console.  sip debug
15:30.47SomeOne1well i wanna keep it on permanently
15:30.53SomeOne1instead of having to type it in console all the time
15:30.55shellsharkyeah man... CLI == friend :)
15:31.03shellsharkdude
15:31.04SomeOne1and see it in messages
15:31.14mercestesIt will be permanent until your server crashes
15:31.17shellsharkyou DONT want that permanently
15:31.23shellsharkmercestes: haha yeah ;)
15:31.45SomeOne1ouch, i see why
15:31.49SomeOne1those are some ugly messages
15:31.56SomeOne1like down to the packet level
15:31.59mercestesuh huh...try beer...it makes them prettier.
15:32.10shellsharklol
15:32.27SomeOne1you guys love nocking on us "nubs" dont you
15:32.31SomeOne1:(
15:32.40SomeOne1jk
15:32.47SomeOne1thanks for the help though
15:32.58mercestesaw...I'm not knocking you.  I'm still a nub.
15:33.00shellsharkSomeOne1: we're not knocking on anyone :p
15:33.08SomeOne1i know im kidding
15:33.19SomeOne1<nays85> so try another provider or another phone number... it could ultimately be the originating PSTN switch where that phone number comes in from
15:33.22mercestesI couldn't write a cascading failover dialplan over 2 colo'd pri's without the wiki.
15:33.24SomeOne1that makes sense
15:33.29mercestesHell, if the wiki went down I'd be out of a job.
15:33.49SomeOne1siterip it
15:33.53SomeOne1or something
15:34.02mercesteslol.  wget wiki.asterisk.org
15:34.04mercestesmuhahahahaha
15:34.06pfhorgesomeone1, they're lying, so much. You make one little comment about ATAs and they're all over you
15:34.08SomeOne1haha yep
15:34.08Rawplayeris there a bot in here?
15:34.17Rawplayerwhat gives usefull output
15:34.17pfhorgethey love to mock us
15:34.24macTijn~kick Rawplayer
15:34.26jbotbugger off, mactijn!
15:34.30Rawplayerk
15:34.30macTijnheh
15:34.32SomeOne1pfhorge, i dont understand, whose "type"... the SIP providers or the people in here
15:34.43SomeOne1err
15:34.46SomeOne1"they"
15:34.46shellsharkmercestes: dial(Zap/1/blah) GotoIF(${DIALSTATUS}='ANSWERED'||${DIALSTATUS}='BUSY'?end) Dial(Zap/2/blah) ?
15:34.47pfhorgemercestes and shellshark
15:34.48SomeOne1not type
15:34.56SomeOne1pfhorge, nah theyre cool
15:35.01SomeOne1i dont take it personally anyway
15:35.11pfhorgeit hurt my feeling.
15:35.12mercestesShellshark:  Show off.
15:35.20mercestesI'm sorry
15:35.39mercesteshey, anyone play with those old "AI" chat robots like "Chat with Lisa?"
15:35.43SomeOne1unless they attacked my personality and race or something... and my physical defeceds
15:35.50SomeOne1like my third arm
15:35.57*** join/#asterisk inspired (n=mikael@62.141.128.222)
15:36.00SomeOne1defects*
15:36.00shellsharkmercestes: how does that make you feel?
15:36.03SomeOne1i have a thiud arm
15:36.24pfhorgeIt's in charge of the 'r' key and it can't even handle that!
15:36.27shellsharki have a third leg, but i doubt one might consider it a defect ;)
15:37.02*** join/#asterisk infernix (i=nix@spirit.infernix.net)
15:37.22shellsharkoooo, silence ;)
15:37.37mercestesWell anyways...I wanna setup a "talk with Lisa" program that uses that text based AI thing.
15:37.40*** join/#asterisk gabb0 (n=gabb0@131.202.90.23)
15:37.59shellsharkmercestes: check out MagaHAL
15:37.59mercestesI can definately catch the output and just festival it out to the user but.....how do I catch what they're saying and feed it back to Lisa?
15:38.13shellsharkmercestes: Sphinx
15:38.25gabb0hello all
15:38.37shellsharksphinx does voice --> text recognition
15:38.39b11d|bblmorning lads
15:38.44mercestesmagahal?
15:38.57gabb0has anyone here setup an ADTRAN TotalAccess 900 series with asterisk before?>??
15:39.02*** join/#asterisk TomWJr (n=twyant@68.76.27.250)
15:39.16TomWJrMornin'!
15:39.36shellsharkgabb0: you doing DSX?
15:40.37SomeOne1guys
15:40.41SomeOne1i see something like
15:40.50SomeOne1a=rtpmap:0 PCMU/8000^M
15:40.50SomeOne1a=rtpmap:18 G729/8000^M
15:40.53*** join/#asterisk kannan (n=kannan@58.68.25.67)
15:40.58SomeOne1is that saying its offering PCMU and G729
15:41.03SomeOne1thats in the invite packet
15:41.42mercestesis it megahal instead of magahal?
15:42.08kannanhello all. I am able to register to a sip server for outbound cals from an Xlite free phone , but asterisk keeps timeout on the registration, any suggestions what do i do next?
15:42.41shellsharkmercestes: i said mega, no?
15:42.42slothgabb0: I have not yet, but I am about to.
15:43.06shellsharkmercestes: my bad ;)
15:43.55mercestes<shellshark> mercestes: check out MagaHAL
15:44.05pifis there a function to remove white space in a var ?
15:44.15pifor a substitution?
15:44.27gabb0shellshark, I'm just getting into looking at it now.  I have a 904 and 924 to  setup.  I have an asterisk box and I want to have the analog lines be fxs stations and connect the two by sip
15:44.33*** join/#asterisk ai-a (n=jake@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)
15:44.47*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
15:44.49ai-awhere can i get info on the fedora code 6 repository for yum installing asterisk ?
15:45.51shellsharkai-a: #fedora?
15:46.37ai-afedora help wont know about asterisk.
15:47.08SomeOne1im gonna try to force it to use g729
15:47.10SomeOne1watch this
15:47.13Qwell[]They'll know about installing it with yum
15:47.33ai-aQwell: i know how to install it, i just need to info on the repository sites.
15:47.52Qwell[]They would know MUCH more about their repository than we would
15:48.07ai-aits not in the fedora repository.
15:48.11QbYin my dialplan, i have a caller who has taken an option that needs me to send them to another telephone number..  however the other telephone number requires me to hit 0 before talking to someone--is it possible for me to script this into the dialplan?
15:48.14Qwell[]You just said it was
15:48.25ai-ayou can create extra repositories of projects.
15:48.43TomWJrAnyone here know how to use conary with business edition?
15:48.48ai-aok... does anyone know where the asaterisk repository is for fedora core 6 ?
15:48.53Qwell[]TomWJr: please call Digium support
15:49.00TomWJrPlease, don't make me
15:49.10Qwell[]TomWJr: You won't get any help with BE here
15:49.19Qwell[]You paid for support - use it :p
15:49.19TomWJrYeah, I figured.  It was worth a shot
15:49.28TomWJrhave you called their support?  I'd rahter call Microsoft!
15:49.44Qwell[]TomWJr: I work with them daily.
15:49.56TomWJrI hope you're getting better help than I am
15:49.59Qwell[]If you are having problems with support, that needs to be dealt with.
15:50.00TomWJrI'll give them a call
15:50.11*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
15:50.12TomWJrOkay, how about this
15:50.43TomWJrDo you (any of you) use the open source version in your installs or do you buy abe for commercial clients?
15:50.56TomWJrI've done abe for all my commercial installs
15:51.02*** join/#asterisk lorinc (n=ang@caracas-1851.adsl.interware.hu)
15:51.07b11dwhat the fuck is abe?
15:51.08TomWJrbut with my support problems I'm not sure I want to
15:51.11TomWJrasterisk business edition
15:51.13b11dohh
15:51.16b11dnever used it..
15:51.21TomWJrit's interesting
15:51.24b11dim sure it is
15:51.27TomWJrfew versions behind the open source version
15:51.36b11doh really??  whats the point of abe then?
15:51.41*** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
15:51.42TomWJrcommercial license
15:51.43TomWJrsupport
15:51.50SomeOne1aha!
15:51.53b11d?/
15:51.54SomeOne1g729 sounds better
15:51.56b11di still dont see the reason :)
15:51.59*** join/#asterisk melllwa__ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
15:51.59TomWJrYeah
15:52.00b11dsupport is fre
15:52.01b11de
15:52.01b11d:)
15:52.08SomeOne1Dec 15 11:52:16 VERBOSE[27624] logger.c: Adding codec 0x100 (g729) to SDP
15:52.08jmlsfellow *'ers: I take it that  ERROR[1843]: chan_zap.c:8135 zt_pri_error: !! Got reject for frame 39, but we have nothing -- resetting!
15:52.11jmlsis a bad thing ?
15:52.11b11dbut yeah.. customers enjoy the peace of mind..
15:52.17TomWJryeah
15:52.28*** join/#asterisk Drazha (n=drazha@208.50.83.133)
15:52.33b11dwhat does abe typically run?
15:52.33TomWJrI've run into a few things with abe that are just making it a nightmare for certain clients
15:52.37b11dpricing, that is
15:52.41ManxPowe1Gads, does NOBODY know about DACS?????
15:52.42TomWJr$700-800 marked up
15:52.45b11dahh
15:52.52SomeOne1is MP3Play() using slin?
15:52.53TomWJrThing is
15:52.55b11dwhat are your issues?
15:52.56Qwell[]TomWJr: see msg
15:53.10SomeOne1because for some reason asterisk is trying to "translate" slin to g729
15:53.10TomWJrI know Qwell[], I know
15:53.17TomWJrI'm venting more than anything
15:53.19TomWJr:)
15:53.19Strom_CManxPowe1: funny, I was just thinking about DACS last night
15:53.28Qwell[]TomWJr: I understand - see the last part of the msg
15:53.30ManxPowe1SomeOne1: do you have a G729 license?
15:53.36SomeOne1ManxPowe1: no
15:53.40SomeOne1will you report me?
15:53.52ManxPowe1SomeOne1: SLN is the internal format Asterisk uses when transcoding.
15:53.53DrazhaCan Asterisk be used with a serial external hardware modem instead of buying the digium cards etc?
15:53.54SomeOne1its just for testing, my SIP origination provider allows others
15:54.00ManxPowe1SomeOne1: No, it just won't work that's all
15:54.01blitzrageSomeOne1: I already have... CIA is on it's way
15:54.10SomeOne1:(
15:54.15blitzrageDrazha: nope
15:54.17SomeOne1its for testing!!!
15:54.19TomWJrAlright, calling Digium
15:54.20TomWJrthanks!
15:54.23ManxPowe1Drazha: NO!
15:54.26*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
15:54.40SomeOne1i <3 asterisk
15:54.43ManxPowe1Strom_C: DACS is the coolest thing since sliced bread
15:54.43SomeOne1i'm gonna make a t-shirt
15:54.55SomeOne1i <3 asterisk, and not a real heart but like text.. in arial font
15:54.59SomeOne1and im gonna wear it around
15:55.08b11dwhat is DACS?
15:55.10blitzrage*coughnerdcoughcough*
15:55.15SomeOne1hahah
15:55.17SomeOne1super nerd
15:55.19Strom_CSomeOne1: if you're going to do ascii art, don't use something as abominable as Arial
15:55.28SomeOne1heh
15:55.34SomeOne1im open to suggestions
15:55.40blitzragefor the nerds: http://www.youtube.com/watch?v=Fow7iUaKrq4
15:55.45ManxPowe1b11d: DACS is a way to cross connect channels using Zaptel outside of Asterisk
15:55.50b11dahh cool
15:56.09b11deveryone go to google video and watch "Haggard" the movie..  
15:56.12Strom_CSomeOne1: a cool monospaced font
15:56.13b11dits so bad its hilarious
15:56.16*** join/#asterisk mega (i=mega@gateway/tor/x-8ae5a68085976a95)
15:56.18ManxPowe1We use it to cross connect channels that don't need call processing like data channels or other things lke that.
15:56.26b11dohh ok.. that makes sense.
15:56.30SomeOne1anyway
15:56.33SomeOne1gotta get back to work
15:56.35SomeOne1thanks for the help
15:56.38b11dthanks for breaking that down for me ManxPowe1
15:56.50Drazhablitzrage like this is the worst thing I saw ever :)
15:56.55b11dit should be illegal to have an attractive boss..
15:57.07*** part/#asterisk mega (i=mega@gateway/tor/x-8ae5a68085976a95)
15:59.03blitzrageb11d: agreed -- unless you're sleeping with her :)
15:59.09blitzrageDrazha: I know... it's just not right :)
15:59.18*** join/#asterisk lters_ (n=tech@mrtcdsl-433.mis.net)
15:59.52Drazhablitzrage its....
16:00.11b11dI wish :)
16:00.33Drazhablitzrage its... ripe for ... I dunno... some terrible thing that will happen that will satisfy and vindicate at the same time, something public and brutal
16:01.46b11dthe worst part is i have a reputation for not caring about marital status :P
16:01.57Drazhaanyway... Can asterix be used with these FXO FXS cards to recieve FAX transmissions etc?
16:02.09blitzrageASTERISK
16:02.15RawplayerDAMNED!
16:02.19blitzrageASS TER ISK
16:02.26blitzragenot ass tricks, not asterix, etc...
16:02.32Drazhaok, obelix
16:02.35blitzrageheh
16:02.38ManxPowe1Drazha: Yes, but in my experience it is less reliable than a direct line to the telco.
16:02.42Qwell[]asterks?
16:02.49blitzrageastjerks?
16:02.56Strom_Ccatsex?
16:02.57Qwell[]You totally are
16:03.02blitzrageQwell[]: I try
16:03.07DrazhaI dont wanna have to buy yet another chunky fax machine thats gonna fall to pieces like after a year of using....
16:03.21ManxPowe1I'm going try to get a vanity license plate that says "ASTMSTR"
16:03.39blitzragefaxing has never been known as asterisks strong point
16:03.40ManxPowe1Drazha: Oh!  No, asterisk does not have a "fax machine built in"
16:03.49pfhorgelooks like my phantom ring only happens on 2 of my Zap channels. Wierd.
16:04.08DrazhaManxPowe1 yeah, I know that. it is friday, it is late, but my brain core functionality is still intact
16:04.11blitzrageyou can pass-through fax with T.38 in 1.4, but that's about it
16:04.27blitzrageor if you have incoming and outgoing directly on the hardware you might be ok
16:04.36ManxPowe1I just get a POTS line for fax/credit card/modem
16:04.39blitzrageif you need to receive fax on the machine, get Hylafax
16:04.55b11dmaybe he should go to a concentration camp!
16:05.04b11dgod dammit!
16:05.06b11d-cartman
16:05.49*** join/#asterisk lters_ (n=tech@mrtcdsl-433.mis.net)
16:05.52ChkDigitExcept it was mice.
16:05.53b11dchixdiggit was a great band
16:06.50tzangerinteresting
16:07.02*** join/#asterisk rsd (n=chaos@200.181.133.130)
16:07.06Drazhaniah, twas keyboard
16:07.07tzangermanhole spacing is 6000ft for load coil requirments with POTS lines
16:07.22tzangerT1 repeater distances were engineered to match this, so they could use the same manholes for repeaters
16:07.51Strom_Cmakes sense
16:07.55*** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it)
16:08.06Strom_CI don't think I have any BSPs that cover T1
16:08.27*** join/#asterisk You_Asterisk (n=younssig@194.204.203.177)
16:08.29tzangerBSP?
16:08.35Strom_CBell System Practice
16:09.24*** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66)
16:09.25Strom_Cfor example...
16:09.47Strom_CSECTION 501-164-204, Issue 1, October 1979
16:09.53irqare there any voip-native fax machines that i can buy to just replace my existing POTS fax machine?
16:10.00irqwe have a need to send faxes, the old fashioned way
16:10.07Strom_C1200AR1 "TOUCH-A-MATIC" 12 ADJUNCT DIAL
16:10.31Strom_Cirq: do you have PRI service coming in from the telco?
16:10.47irqStrom_C: no. analog T1 unfortunately.
16:11.00Strom_CT1 isn't analog :)
16:11.18irqparts of non-PRI-T1 signaling are indeed analog
16:11.40Strom_Csure, the inband signaling is analog, but the actual carrier itself is digital and has been since the day T1 was invented
16:11.49kannanhello, anyone can suggest what do to about registration timeouts on asterisk ? i am able to use the sip account with an xlite softphone
16:11.53irqthe fax machine we have now works okay over our analog->voip adapter, but i'd love to just have ethernet straight into the fax machine. do you know of any fax machines that do that?
16:13.11Strom_Cirq: the best way to do fax in that environment, honestly, is to get a channel bank on another span of your multi-span T1 card, hook the traditional fax into that, and just bridge the two across the same T1 card
16:13.27Strom_Cand I'm not aware of any such thing as a voip fax machine
16:13.46irqunfortunately that's not an option without really changing our voip setup, because right now my T1 goes straight into a digium card with no hardware in between, and i really like it that way
16:13.58irqand our t1 card has just one port
16:14.07Strom_Cirq: ah, yeah.  multi-span t1 card for the win.
16:14.21irqso, i won't be doing any of that
16:14.31irqthe cost / pain involved isn't worth the minor inconvenience of our fax having problems occassionaly
16:14.39irqso there is no voip native fax machine?
16:14.50Strom_Ca client who I set up with the channel bank hasn't had any problems ever, and the client who insisted on the terminal adapter does nothing but complain about fax reliability
16:14.59Strom_Cirq: no, not as far as I'm aware.
16:15.15irqi'm not saying we won't have problems, i'm saying that management isn't going to go for it and authorize the funds
16:15.35wunderkinso... don't complain?
16:15.38DrazhaI imagine an IP based fax machine aint gonna be chep either
16:15.44Strom_Cirq: I understand; i'm giving you ammo to battle management with :)
16:15.45*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
16:15.58irqwunderkin: my job is to bridge the whiney users to the executives :)
16:17.12ManxPowe1kannan: the way to fix registration timeouts is to fix your network
16:18.44*** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net)
16:21.45*** join/#asterisk melllwa_ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
16:23.12*** join/#asterisk sil (n=sil@terrorist.infiltrated.net)
16:25.03b11dI wonder if I should move up to the 1.2 branch of Asterisk or not.
16:25.36b11d...
16:25.39b11dnothing?
16:25.42b11dok im kidding..
16:25.54*** part/#asterisk Drazha (n=drazha@208.50.83.133)
16:26.43caio1982the build mechanism of 1.2 (comparing to 1.4's) is my very first reason to never use 1.2 again :P
16:27.18b11dI havent even run 1.4 yet..
16:27.29b11dany word on when its going to be non-beta?
16:27.33mercestesI did...I wasn't impressed.
16:28.00caio1982just natural evolution, nothing that will save the world
16:28.17kannanManxPowe1 : I have ping the sip proxy and also made calls with the xlite phone , but asterisk will timeout
16:28.30*** join/#asterisk ruffle (n=russell@ipcop.llsnet.co.uk)
16:28.32b11dwhat werent you impressed with mercestes?
16:28.58*** join/#asterisk Gamentine (i=WinNT@S0106000d61588a55.cg.shawcable.net)
16:29.07*** part/#asterisk Gamentine (i=WinNT@S0106000d61588a55.cg.shawcable.net)
16:29.25mercestesI dunno...I expected laserbeams and the solutions to all my problems.
16:29.40b11dnatrually.. the changelog shows laser beams were added..
16:29.45mercesteswhat I bot was 1.2.13 that wasn't guaranteed to stay running.
16:30.21mercestesI was mainly annoyed that ChanIsAvail was still busted.
16:30.27mercesteswhich is the only reason I upgraded.
16:30.29b11di guess i havent seen any issues with 1.2.13 and stability..
16:30.37mercestesIt errored differently..mind you...
16:30.46b11dI'm looking forward to the enhanded MoH in 1.4
16:30.52b11denhanded = enhanced
16:31.08caio1982what you mean by enhanced?
16:31.10sil1.2.13 has been giving me issues clearing channels
16:31.23ruffleCan anyone give me a clue why incall DTMF features are not being acted on in Asterisk 1.4.0b3 ?  Asterisk sees the dtmf-relay SIP message but doesn't do anything about it :(
16:31.34fileruffle: fixed in the 1.4 branch
16:31.46fileb11d: everyone's deployments are different, so people see different issues
16:32.06ruffleAh. OK will try that. Thanks.
16:32.07sil1.2.13 hasnt been closing the channels so i had to make a script running from cron to close channels still open
16:32.17filesil: what kind of channels?
16:32.27silsip channels
16:32.32mercestesThat reminds me...i need to finish my 1.4 beta installation so I can test the chanisavail for the bugreport I submitted on it..:/
16:32.43sil... /usr/sbin/asterisk -rx "show channels concise" | awk -F : '($11 > 5400) {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh does it for me
16:32.55b11dyep.. you're right file..
16:32.57*** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee)
16:33.00silkeeps channels opened long after callers have hung up
16:33.10filesil: what kind of channels though?
16:33.34*** join/#asterisk mega (i=mega@gateway/tor/x-c21285dc7f7ff2a8)
16:33.52kannanhowto spoof the xlite in asterisk , i have changed the useragent to the xlite UA
16:33.52b11dcaio1982.. i understand MoH is a lot more configurable in 1.4 -- like, with the ability for end users to define their own MoH..
16:33.55b11di might be wrong
16:34.00silsip channels file
16:34.22filesil: have you done a sip debug to confirm the SIP dialog, turned on sip history?
16:34.23siltheyre sip channels the call goes away on... it never receives any indication the call is gone like the wind
16:34.41silyea i got a separate log entry in logger.conf
16:34.57silall sipdebugging goes to /var/log/asterisk/sipdebug
16:35.20silnot a big deal to me i know how to kill them... might not be good for people charging per minute though
16:35.47mercestesI did find an issue where if you had an established SIP call and unplugged the phone that it would hang a sip channel.  Could be a similar issue.
16:36.20*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
16:36.29filemercestes: that's the way it works unless you have the RTP timeout stuff enabled, and if your audio stream is off the server you would never know
16:38.10mercestesYea, that's what I was gathering.
16:38.45*** join/#asterisk melllwa__ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
16:38.59b11dso whats cooking this weekend for everyone?
16:39.04b11dno work I hope
16:39.36rudholmthis weekend I hope to make some time for my asterisk coin phone service project. :)
16:39.50b11dneato :)
16:40.00b11dI've got a few old cocots..
16:40.02b11dlets modify them
16:40.18rudholmI have a "real" payphone
16:40.32b11dsame.. lifted it from Bell Canada back in Ontario
16:40.35Strom_Ccocots are for the birds
16:40.38b11dfuckers are heavy
16:40.41rudholmright now it's connected to a telco Coin Service line, but I want to connect it to Asterisk
16:40.45b11dyeah but cocots were easy to hack
16:40.54Strom_Cb11d: blah blah blah, they still suck
16:40.57b11di dont disagree
16:41.09Strom_CCO-controlled ftw
16:41.21rudholmyeah, with a COCOT, I could connect it to Asterisk no problem, since they're designed to be connected to regular (i.e. not coin service) lines.
16:41.36b11dI just like saying "cocot"
16:41.40rudholmhehe
16:42.04b11din an unrelated topic, i found a bunch of condoms in a public kiosk here on campus today
16:42.06b11dthat was nice..
16:42.12rudholmI think I like "Asterisk Serviced Coin Operated Telephone" better --  "ASCOT"
16:42.13b11dsomeone busted out a fan on the side and started dumping them in
16:42.23b11dASCOT has a nice sound to it
16:42.54rudholmthe problem with free condoms is they tend to be the cheaper ones.
16:43.06rudholmnot the good/expensive polyeurethane ones.
16:43.07b11dI am unaware of the condom's source..  just its destination
16:43.13b11dwhich is the kiosk :)
16:43.54b11dim going to burn that kiosk where it stands..
16:44.00*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
16:44.02fileI sometimes question the sanity of this channel
16:44.07b11dits my fault..
16:44.12b11dban me if necessary
16:44.13b11d:P
16:44.18Strom_Cfile: I'm amazed you thought it had sanity to begin with
16:44.21Qwell[]file: ...some...times?
16:44.27fileStrom_C: I know, I'm a fool
16:44.51fileStrom_C: dogballs!
16:44.54b11dkatty selling that shit on ebay again?
16:45.00robl^catsex!?!?!  not sure I want to know...
16:45.04mercestes...omg..what do I search to make bids??
16:45.08*** join/#asterisk jmesquita (n=jmesquit@201.7.117.114)
16:45.12Strom_Cno no no, you're supposed to reply "dead hookers"
16:45.23Qwell[]not dogballs?
16:45.28mercestesdead hookers.....are no fun at all.
16:45.29b11dI wonder if "dead hookers" is is oxymoronic
16:45.35Strom_Cfile already said dogballs
16:45.37rudholmyeah, they're just a liability
16:45.38Qwell[]b11d: it's redundant
16:45.41Qwell[]Strom_C: ahh
16:45.41b11di thought so
16:46.46b11dI need to quit my job..  cant take it anymore..
16:47.13rudholmwhy?
16:47.15rudholmwhat's the job?
16:47.17b11dhot boss
16:47.19b11dmarried
16:47.21b11dcant take it :P
16:47.25rudholmah
16:47.37b11dnothing to do with the workload :)
16:47.38rudholmthat's not so bad
16:47.39rufflefile: Just tried Asterisk SVN-branch-1.4-r48487 and the SIP dtmf-relay is still being ignored. Any other ideas?
16:47.42b11dno its not so bad..  
16:47.45*** join/#asterisk eddi (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
16:47.55*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-246-217-92.washdc.fios.verizon.net)
16:47.55b11dits just fashionable here in the USA to have something to bitch about
16:48.02rudholmheh
16:48.02SomeOne1how can i get asterisk to make calls from the command line
16:48.07SomeOne1or through a script
16:48.10SomeOne1or from cli>
16:48.12Strom_CSomeOne1: call files
16:48.17Strom_Cor the Dial command
16:48.19b11dsource?  you guys scientologists?
16:48.29Strom_Cwtf?
16:48.40fileb11d: so it's when using the features stuff?
16:48.49b11dwhat?
16:48.52SomeOne1strom_c, can i do something like... dial(blah) and mp3play(/blah.mp3) from cli?
16:48.55Strom_Chow the hell does "source code" imply anything to do with l rom hubbard?
16:49.02Strom_CSomeOne1: no
16:49.03b11dhaha its just "source"
16:49.13b11dthey claim that their knowledge comes from "source"
16:49.16b11dor they call it "source"
16:49.17b11dand "tech"
16:49.24Strom_Cthey're a bunch of nutjobs
16:49.28rudholmI haven't tried it, but can't you just create a call file as well?
16:49.39b11dyes, they are..
16:49.41filestrange strange people..
16:49.43b11dever see that Xenu TV shit?
16:49.43b11dwow..
16:49.49HarryRAnybody tried compiling asterisk with Intel's ICC?
16:49.50caio1982have you checked the originate commando?
16:49.53SomeOne1rudholm: i guess
16:50.14b11dthis is my favorite: http://www.xenutv.com/mb/revenge.htm
16:50.22b11dscientologists picketing a guys house
16:50.24rudholmSomeOne1: sort of like injecting an email into an MTA's mail queue :)
16:50.26b11dpoor bastard
16:51.05*** join/#asterisk copantl (n=galel@190.4.22.94)
16:51.18SomeOne1heh
16:51.21copantlhello
16:51.23b11dhio
16:51.36Strom_Colleh
16:51.48b11dStrom is a sweet name..
16:51.50copantlany body know how to change the hangup cause to cause 34?
16:51.53SomeOne1rudholm: i can run like MP3Player() from a .call file?
16:52.03Strom_Cb11d: thanks - it's a phone joke
16:52.07*** join/#asterisk stephane (n=stephane@gw.sortilege.net)
16:52.11b11doh really? thats cool
16:52.22SomeOne1or set(callerid)
16:52.22SomeOne1?
16:53.19rudholmSomeOne1: dunno, I haven't done it, just read that it can be done.
16:53.23ManxPowe1copantl: I believe HANGUPCAUSE is read only
16:53.30Strom_Cno, you can set hangupcause
16:53.44ManxPowe1I sit corrected
16:53.47copantlis a function right?
16:54.03copantlhow can i sen cause 34 to other party?
16:54.10Strom_Cin 1.4, you specify the cause code as an argument to Hangup()
16:54.23*** join/#asterisk melllwa_ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
16:54.23Strom_Cin 1.2, IIRC, it's the HangupCause() app
16:54.24ManxPowe1copantl: OH!  Why didn't you say so in the first place!
16:54.53ManxPowe1copantl: "show application hangup"  "show application busy" and "show application congestion"
16:54.55*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
16:55.09copantlthank a lot
16:55.22Strom_Cahhh ok
16:55.23Strom_Chang on
16:55.31Strom_Cyou set PRI_CAUSE in 1.2 and earlier
16:55.42Strom_CSet(PRI_CAUSE=34)
16:55.43Strom_CHangup()
16:56.22copantlStrom_C: thats it?
16:56.28Strom_Cyes
16:56.34copantlis just set the cause?
16:56.49copantland them hangup
16:57.00Strom_CHangup() will hang up and send the cause code set in PRI_CAISE
16:57.03Strom_Cer, CAUSE
16:57.09copantlwhere i set de PRI_CAUSE
16:57.28Strom_C?
16:58.13*** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com)
16:58.13copantlwhere i put Set (PRI_CAUSE=34)?
16:58.19copantlin the dial plan?
16:58.39Strom_Cjust before Hangup(), just like I just showed you
16:58.43copantlok
16:58.52copantlthanx a lot Strom_C
16:58.56*** part/#asterisk mega (i=mega@gateway/tor/x-c21285dc7f7ff2a8)
16:59.00Strom_Cmerry dogballs
16:59.14copantloh, another thing, it work in all versions?
16:59.25Strom_Cdude, did you not listen to me at all?
16:59.33Strom_C1.2 and earlier, set PRI_CAUSE
16:59.43SomeOne1oklay i made a .call file
16:59.45SomeOne1how do i run it?
16:59.51Strom_C1.4, pass the cause as an argument to Hangup()
16:59.54copantlsorry
17:00.05Strom_CSomeOne1: copy it to /var/spool/asterisk/outgoing/
17:00.19MrChimpysomeone: alternatively rtfm
17:00.33b11dthe good old standby of RTFM..  I love it
17:00.44b11dI've got a (im not proud) RTFM mug I walk around the school with..
17:00.55MrChimpyit's not exactly an undocumented feature nor a tricky one
17:00.57Strom_CRTFM would make a killer license plate
17:00.58*** join/#asterisk _alex_mx_ (n=alex@200.78.229.18)
17:01.06SomeOne1b11d: are you a teacher?
17:01.07b11dyeah it would.. I think they know about that stuff now though
17:01.17b11dnah, im the system admin of a school
17:01.24SomeOne1high school?
17:01.35b11dthey want me to teach but only as an "adjunct" instructor -- which means 1/3rd the salary and the same amont of work as regular teachers.
17:01.37b11dno, a college.
17:01.46copantlStrom_C: in 1.4 can be like this Hangup(34)?
17:01.52ManxPowe1b11d: you turned them down of course
17:01.55b11dhell yes
17:01.57Strom_Cthat's what I just said
17:02.02b11dthey can go fuck themselves
17:02.13ManxPowe1for some reason I thought 1.2 supported Hangup(CAUSECODE)
17:02.13SomeOne1ahh, minnesota
17:02.16b11dyep :)
17:02.22Strom_CManxPowe1: it may
17:02.26*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:02.27ManxPowe1What IS cause 34?
17:02.32SomeOne1hows Keith Ellison
17:02.37Strom_Ci know the previous version supported the pri_cause thing
17:02.45copantlManxPower1: is congestion
17:02.46rudholmyeah, life is kinda bumming until you get (or are within sight of) full tenure
17:03.24ManxPowe1copantl: just run Congestion in your dialplan then
17:03.29ManxPowe1it wil send a cause 34
17:03.35Strom_Cah dammit, someone already has california license plate "RTFM"
17:03.35b11dEllison is causing a big stir over NOTHING here..
17:03.43b11din fact, its not even him.. its others..
17:03.52copantlMandPowe1: but i need to send cause 34
17:04.03rudholmStrom_C: I'm surprised they got that one to slip through.  the CA DMV is pretty vigilant about the meanings of designations.
17:04.11copantlnop  can send cause 16
17:04.16rudholmthey must must have come up with a plausible alternative definition
17:04.23copantland i just need to send cause 34
17:04.25ManxPowe1copantl: Congestion() should send a CAUSE 34
17:04.29Strom_CManxPowe1: does congestion automagically release the channel on PRI?  I don't recall
17:04.51ManxPowe1Strom_C: it should
17:05.00copantlbut i need it everitime
17:05.07ManxPowe1If it does not then hangup will
17:05.23b11dknow whats the worst part about working at a college?  I'm only 25..  the girls here are my age..  and im prohibited from enjoying a "relationship" with them..
17:05.24b11dargh..
17:05.26Strom_Ccopantl: you're now in the semifinal round of the 2006 "not listening" awards
17:05.28ManxPowe1copantl: Doing a Hangup(34) in 1.4 would be exactly the same as Congestion()
17:05.57copantlsorry guys..but english is not my first languages
17:06.15*** join/#asterisk melllwa__ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
17:06.24rudholmb11d: really?  I don't know of any other schools that have that policy except for very direct conflicts of interest (e.g. student-teacher, student-lab assistant, etc)
17:06.46b11dits because im the system admin and have damn near full access to the statewide record system..
17:06.49b11daka.. can alter grades..
17:07.10b11di think its an extreme policy too though..
17:07.19b11din fact.. I really should look that up and make sure people arent just BSing me
17:07.26rudholmone would think the db would have an audit trail
17:07.42b11dit does, but those trails are only examined when things arent going well..
17:07.43SomeOne1b11d: what do you think of ellison?
17:07.50rudholmgranted, sysadmins can modify an audit trail, but you could make it pretty difficult.
17:07.57rudholmI see
17:07.57b11dI have no particular opinion of him.. I say let him take his oath on the Koran though..
17:08.02b11dwho cares if its the holy bible or not..
17:08.10SomeOne1god damnit i can receive calls from my SIP provider but i cant make calls from behind my NAT, they get screwed up
17:08.13b11dits all about his WORD and he believes in Islam so he should take it on the Koran
17:08.25SomeOne1b11d: cool, me too
17:08.27SomeOne1thats what i think
17:08.32ManxPowe1SomeOne1: you, of course set externip= and localnet=
17:08.40SomeOne1heh
17:08.41SomeOne1yeah
17:08.43SomeOne1and nat=yes
17:08.45b11dright on..  
17:09.05b11dBut as for Ellison as the man or as the congressman..  time will tell..
17:09.12SomeOne1true
17:09.13ManxPowe1b11d: I have a radical idea.  How about we remove religion from the swearing in of a govt official!
17:09.23b11dTHIS IS AMERICA DAMMIT!!
17:09.25SomeOne1ManxPowe1: great idea!
17:09.25b11dYOU CANT DO THAT!!
17:09.26b11d:)
17:09.27b11dhaha
17:09.31SomeOne1heh
17:09.32SomeOne1yeah
17:09.39ManxPowe1SomeOne1: Paste the Dial line
17:09.47luke-jr|workManxPowe1: wtf?
17:10.04b11dYeah religion needs to be removed from those offices..  however, realistically, its not reasonable to expect a true seperation of church and state.
17:10.10b11dits not humanly possible
17:10.18luke-jr|workSeperation of church and state is heresy.
17:10.23b11dpeople WILL make decisions based on their faith.. pure and simple.
17:10.28ManxPowe1b11d: Perhaps not, but the can at least TRY
17:10.40b11dI agree.. unfortunatly most people just "omit" instead of trying..
17:10.45rudholmand you can at least acknowledge that religion has no place in the laws of a secular state.
17:10.54b11dhaha
17:11.02b11dyeah.. agreed :)
17:11.06luke-jr|worknonsense :)
17:11.08rudholmthis *is* a secular state
17:11.16b11dwe got the joke
17:11.17ManxPowe1The House and Senate start each session with a PRAYER
17:11.25rudholm...now.
17:11.27luke-jr|workrudholm: actually, it's a protest-ant state :p
17:11.33rudholmheh
17:11.52*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.6)
17:11.55kippiif I just want to make the extenstions ring and ring and nerver stop until someone picks up what would I want to put at the end of my dial command?
17:11.55b11dI have no particular issue with prayer as long as all the people involved practice the religion upon which all good people can agree.
17:11.59b11dwhich I like to call "common sense"
17:12.16*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.6)
17:12.19ManxPowe1kippi: NOTHING
17:12.40luke-jr|workb11d: might want to call it something else
17:12.45b11dmaybe..
17:12.47luke-jr|worka lot of people misinterpret "common sense"
17:12.52kippiI've put that, the phones ring and ring and keep on ringing but my mobile phone stops
17:12.54b11dwho's going to the 2008 RNC in Minneapolis??
17:12.57b11dim there for sure
17:13.03b11dluke-jr.. you are correct..
17:13.06ManxPowe1kippi: then your MOBILE is stopping it
17:13.15SomeOne1ManxPowe1: why cant i make outgoing calls from behind my NAT :'(
17:13.22rudholmkippi: your mobile carrier probably doesn't allow indefinite ringing
17:13.33rudholmkippi: I've never seen one that does
17:13.35kippiah ha, how comes the phones keep on ringing after my phone cuts off, can i put a stop to it?
17:13.39ManxPowe1SomeOne1: I might be able to help you but you totally ignored my request to paste your Dial line to the channel
17:13.55*** join/#asterisk renato_ (n=v0id@20150185197.user.veloxzone.com.br)
17:14.11ManxPowe1kippi: perhaps you did something stupid and put the "r" option on the Dial line?
17:14.32rudholmkippi: I'm not sure what you mean.  you've used the word "phones" and "phone" somewhat ambiguously.
17:15.19*** join/#asterisk avalone (n=avalone_@dial-191.vl-cen-as2.avtlg.ru)
17:15.19*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
17:15.40ManxPowe1The "r" option to Dial means (r)emove all ability to tell what is really going on with the call
17:15.43hoobastoobaas asterisk forks new processes under load, should those processes go away when they are not in use?
17:16.07hoobastoobaasterisk creates 1 or two new processes every day, but I have 3 days worth in my top list
17:16.28ManxPowe1hoobastooba: Asterisk does not fork new processes under load.  It may spawn a new THREAD.  Different versions of "ps" and "top" show this differently.  some show threads as seperate processes, some don;t
17:16.38b11dtop sucks
17:16.39SomeOne1Channel: SIP/5712417594@icall
17:16.39SomeOne1CallerID: Adeel Mufti <7032817860>
17:16.39SomeOne1MaxRetries: 0
17:16.39SomeOne1RetryTime: 0
17:16.39SomeOne1WaitTime: 500
17:16.40SomeOne1Application: MP3Player(/jesus.mp3)
17:17.17hoobastoobayeah, thats probably what it is... but shouldnt they go away after everything goes idle?
17:17.18ManxPowe1SomeOne1: Why don't you try something SIMPLE to troubleshoot this issue, like a normal phone call.  Also for more than 2 lines of paste, use pastebin.ca
17:17.39SomeOne1whats a pastebin.ca?
17:17.40ManxPowe1hoobastooba: they don't use up any more memory or resources
17:17.41b11dgo there
17:18.09ManxPowe1~pb
17:18.10jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
17:18.15b11dbefore "zee germans" get here
17:18.33hoobastoobaWhat I have been getting are multiple lines of: "/usr/sbin/asterisk -vvvg -c"
17:18.49hoobastoobaevery time I kill one of them i get more memory back in my top summary.
17:18.59*** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net)
17:19.13ManxPowe1hoobastooba: are you using AGI or something like that?
17:19.19hoobastoobayeah
17:19.40ManxPowe1that would do it if the AGI is not catching the correct signals
17:19.50SomeOne1ManxPowe1: in the sip debug log everthing seems all fine but all of a sudden it schedules destruction of call
17:20.05SomeOne1for no aparent reason
17:20.20ManxPowe1when a channel that is running an AGI hangs up Asterisk sends the AGI a signal (SIGHUP?) and if the AGI does not trap that and exit it will stay around forever
17:20.21kippirudholm: ok, now with just the extenstions in the Dial my phone keeps on rining but the handsets stop runing, is there system wide thing?
17:20.46ManxPowe1kippi: what does it show on the CLI?
17:20.54hoobastoobaManxPowe1: interesting... how can I better trouble shoot that?
17:21.14ManxPowe1hoobastooba: do you see Zombies in your PS
17:21.27hoobastooba1
17:21.47hoobastoobawell... i see it in top
17:21.57*** join/#asterisk dmdnb9s (n=dmdnb9s@c-24-20-35-49.hsd1.mn.comcast.net)
17:22.01*** join/#asterisk Zand3r (n=Zand3r@spc1-bolt7-0-0-cust660.bagu.broadband.ntl.com)
17:22.20ManxPowe1hoobastooba: try "ps -ax"
17:22.22kippinothing is shown, just that the extenstions are ringing
17:22.29SomeOne1ManxPowe1 :(
17:22.38ManxPowe1kippi: so you see Dial being executed and then nothing else?
17:22.43kippiyeah
17:23.02ManxPowe1SomeOne1: until you simplify things you'll never get the problem fixed.
17:23.10*** join/#asterisk [Airwolf] (n=airwolf@84.241.200.213)
17:23.19hoobastoobaManxPowe1: STAT = Z would be zombie, right?
17:23.34ManxPowe1SomeOne1: pick up a phone connected to Asterisk. Call someone.  Paste the dial line used.
17:23.38ManxPowe1hoobastooba: I believe so
17:23.56copantlStrom_C: i did this : exten => _X.,1,Dial(ZAP/g0/${EXTEN:7}) and them exten => _X.,2,Congestion()
17:23.59hoobastoobaso i do have one listed, which matches what top tells me.
17:23.59ManxPowe1hoobastooba: just remember that if a process isn't doing anything it may not be listed in sop.
17:24.01hoobastooba4544 ?        Z      0:00 [sh] <defunct>
17:24.27*** join/#asterisk melllwa_ (n=melllwa@HSI-KBW-085-216-067-042.hsi.kabelbw.de)
17:24.39Strom_Ccopantl: what's on g0?  a PRI?
17:24.42copantlStrom_C: but dont change de hangup code ... i did a test and send cause 17
17:24.47SomeOne1ManxPowe1: mayve my router doesnt support rport?
17:24.48copantlyes
17:25.02Strom_Ccopantl: well that's not going to work regardless
17:25.02copantlPRI/ISDN
17:25.39Strom_Cbecause if the other end supervises and then unsupervises, the channel will tear down and never get to priority 2
17:25.45ManxPowe1SomeOne1: Asterisk's nat does not require ANY router support.  In fact if a router has special support for SIP+NAT it will break Asterisk's support for SIP+NAT
17:25.50*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
17:26.11hoobastoobaManxPowe1: so are you suggesting that my problem is dialplan related or possibly my php script called by the agi?
17:26.36ManxPowe1hoobastooba: php  There was talk about zombie php AGIs several times on the mailing lists
17:27.01kippiManxPowe1 Nothing else apart from ringing it beeing shown
17:27.02ManxPowe1kippi: what type of phone are you using?
17:27.04hoobastoobavery good... on my way there.
17:27.08ManxPowe1to make the call?
17:27.51*** join/#asterisk converx (n=locid@206-248-176-51.dsl.teksavvy.com)
17:27.52ManxPowe1kippi: put the entire CLI output of a call on pastebin.ca
17:28.02copantlStrom_C: dont understand
17:28.26ManxPowe1copantl: if the other end answers then Nothing else will be processed in the dialplan
17:28.27SomeOne1ManxPowe1: Dec 15 13:16:17 WARNING[27962] chan_sip.c: Maximum retries exceeded on transmission 4590eb21573925b52c1b59a410ecd331@71.246.217.92 for seqno 102 (Critical Request)
17:28.46*** join/#asterisk saftsack (n=saftsack@pD9E07D8E.dip.t-dialin.net)
17:28.47SomeOne1it means that basically like, it tried sending the "critical request" (seqno 102) like so many time
17:28.48converxin queue cmd,  is it possible to specify how many rings each member receives?
17:28.52SomeOne1and didnt get a reply
17:29.02ManxPowe1SomeOne1: You are about to get the "Doesn't Listen" award.
17:29.12SomeOne1ManxPowe1: im being as simple as possible dude
17:29.21SomeOne1sorry man
17:29.22SomeOne1heh
17:29.31copantlManxPowe1: i need the hangup()
17:29.32Strom_Cconverx: no, but it is possible to specify how many seconds each member is alerted for
17:29.45ManxPowe1SomeOne1: no you are not.  Being as simple as possible is picking up a phone connected to asterisk, making a call to confirm the problem, then putting the Dial line used on pastebin.ca
17:29.59Strom_Ccopantl: you can't send congestion after you've sent ANSWER
17:30.00ManxPowe1If your dial line is not correct then Asterisk will NEVER to special NAT support.
17:30.38*** join/#asterisk navigo (n=navigo@adsl-230-134-184.gnv.bellsouth.net)
17:30.40*** join/#asterisk slayer192 (n=slayer19@66.138.39.225)
17:30.43b11dyeah
17:30.44ManxPowe1SomeOne1: Max retries exceeded just further indicates a problem with the Dial line
17:31.03ManxPowe1Are you afraid of your Dial line?  It doesn't bite.
17:31.06copantlStrom_C: but if i do that can cause false answer?
17:31.21Strom_Ccopantl: what the hell are you asking?
17:31.43*** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it)
17:31.54SomeOne1lol
17:32.00*** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net)
17:32.12ManxPowe1Well kids, we've run out of time for today.  Better luck tomorrow
17:32.17copantlStrom_C: i just need to send congestion every time the call can reach the other party
17:32.17SomeOne1it seems that it never receives an ACK after an INVITE is sent to the SIP server outside my NAT
17:32.46SomeOne1so the outside server isnt able to send back an ACK to my asterisk inside the NAT
17:32.59Strom_Ccopantl: so make your Dial() command time out and then fall through to Congestion().
17:33.04copantlStrom_C: why?: because im connected to a cisco
17:33.22copantland the cisco needs to reicive code 34
17:33.45Strom_Ccopantl: so make your Dial() command time out and then fall through to Congestion().
17:33.52*** join/#asterisk qdk (n=qdk@0xc213c3df.inet.dsl.telianet.dk)
17:39.37b11dlook, when I throw a dog a bone, I dont want to know if it tastes good or not.. and if you stop me when im walking again i'll be sure to cut your fucking minerals off
17:40.13b11dahh Bricktop.. the only person I aspire to become
17:42.51*** part/#asterisk ming_zy2 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
17:45.40b11dso.. any word from the dev's about when 1.4 might be non-beta?  
17:46.16fileeveryone time someone asks a day gets added.
17:46.19fileerm]
17:46.20fileevery time
17:46.35b11dI think I was here when you said that the first time
17:46.37b11dso.. any word from the dev's about when 1.4 might be non-beta?  
17:46.40b11d:P
17:46.42b11dok im done..
17:46.57caio1982so it will be release circa 2009
17:47.01caio1982:D
17:47.07b11dhaha
17:47.08caio1982released*
17:47.11Qwell[]he said day, not hour
17:47.16caio1982haha
17:47.19rufflefile: Any further suggestions about dtmf-relay SIP features getting ignored? I've tried the latest svn; no difference :(
17:47.26b11dbbl lads
17:47.35*** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)
17:47.50fileruffle: pastebin the console output, your dial statement, and add dtmf to the console in logger.conf
17:48.35*** join/#asterisk sloth (n=sloth@64.3.170.41.ptr.us.xo.net)
17:49.21*** join/#asterisk seele_ (n=seelen@dns.datawareltda.com)
17:49.30*** join/#asterisk sloth (n=sloth@64.3.170.41.ptr.us.xo.net)
17:50.04seele_please help how can I recover a call that rings in other extension?
17:50.14Qwell[]seele_: run over to the phone and pick it up
17:50.28caio1982quickly! go!
17:50.35rufflefile: OK, Dial is in a std_sip macro: exten => s,1,Dial(${ARG2},10,Trt)                       ; Ring the interface,
17:50.35*** join/#asterisk jking2100 (n=jking@ool-4351e0f7.dyn.optonline.net)
17:50.45seele_Qwell, no really .... I need this
17:51.24ruffleAhh.. adding dtmf to the logger shows "[Dec 15 17:51:23] DTMF[20393]: channel.c:2128 __ast_read: DTMF end '*' received on SIP/112-0070a6e0"
17:51.35rob0Too late, the caller hung up. :(
17:52.19fileruffle: and that is what you pressed?
17:52.59rufflefile: Yes, I expected it to Disconnect the call as that's what features is set to.
17:53.16filewhat if you hit #
17:53.49rufflefile: [Dec 15 17:53:57] DTMF[20414]: channel.c:2128 __ast_read: DTMF end '#' received on SIP/112-b4800a00
17:54.12filebut no transfer?
17:54.22ruffleNope. the call is still connected
17:54.30*** join/#asterisk ]Airwolf[ (n=airwolf@89.205.138.93)
17:54.40*** join/#asterisk RoyKa (n=roy@217-175-39.100710.adsl.tele2.no)
17:55.15ruffleFWIW, '#' is shown as "Blind Transfer" and '*' as Disconnect Call when I do show features
17:55.35*** join/#asterisk sloth_ (n=sloth@64.3.170.41.ptr.us.xo.net)
17:56.59fileset debug output to go to console, and do core set debug 9
17:57.07fileand also a core show version to confirm the revision
17:57.25ruffleAsterisk SVN-branch-1.4-r48487 built by root @ asterisk on a x86_64 running Linux on 2006-12-15 16:42:24 UTC
17:57.28ruffleis the version
17:57.36filek
17:58.07ruffleHere's the console o/p
17:58.10ruffleasterisk*CLI> core set debug 9
17:58.11ruffleCore debug was 0 and is now 9
17:58.11ruffle<PROTECTED>
17:58.11ruffle<PROTECTED>
17:58.12ruffle<PROTECTED>
17:58.12filepastebin
17:58.12ruffle<PROTECTED>
17:58.15ruffle<PROTECTED>
17:58.17ruffle[Dec 15 17:58:06] DTMF[20433]: channel.c:2128 __ast_read: DTMF end '*' received on SIP/112-b4800a00
17:58.19ruffleasterisk*CLI>
17:58.39file1. Use pastebin next time, 2. Debug output is not going to your console
17:58.59ai-acat /var/log/messages >#asterisk
18:01.17rufflefile: Sorry.. how do I set the debug output to go to the console. I can do core set debug channel ... is that what you meant?
18:01.35fileyou have to add it in logger.conf to go to console, as you did dtmf
18:02.00ruffleAh right oh. sorry for being a thicky. Back in a mo then.
18:02.29*** join/#asterisk wasim (n=wasim@203.81.230.108)
18:05.11Kattyi'm full of pizza.
18:05.48Kattyfile: i'm all cheesy.
18:05.48fileKatty: eep
18:06.14rufflefile: OK I'm a numpty. My IRC client doesn't have a pastebin (Ksirc) but here's the console o/p with debug when I press * in a call
18:06.15ruffle[Dec 15 18:04:08] DTMF[20535]: channel.c:2128 __ast_read: DTMF end '*' received on SIP/112-00704f60
18:06.15ruffle[Dec 15 18:04:08] DEBUG[20535]: channel.c:3726 ast_generic_bridge: Got DTMF begin on channel (SIP/112-00704f60)
18:06.16ruffle[Dec 15 18:04:08] DEBUG[20535]: channel.c:3990 ast_channel_bridge: Bridge stops bridging channels SIP/112-00704f60 and SIP/114-0070afb0
18:06.24file~pb
18:06.25jbot[pb] a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
18:06.32Kattyi sure could use a name of an apt-gettable jabber server.
18:06.44Kattyso if anyone just happens to have a name, back in the corner of their dusty brains..
18:06.51Kattyhmmhesays: like you, maybe.
18:06.54Kattyanthm: or you.
18:06.58Kattythat'd be....swell.
18:08.05fileruffle: I think you did find a bug now that I think about it... can you please file a bug report on bugs.digium.com?
18:08.40lters_Katty: apt-cache search jabber
18:09.01ruffleOK will do. Thanks for the help.
18:09.58b11ddammnit.. I hate when co-workers dig out six year old photos of me getting drunk at a baseball game..
18:09.59*** join/#asterisk hbsmurf (n=twyant@68.76.27.250)
18:10.06b11dand then spread that photo around like its the funniest thing in the world
18:10.19b11dso i've got a few beers..  whats the big deal
18:10.29hbsmurfGot a few beers or HAD a few beers?
18:10.35b11dmaybe a little of both
18:10.42hbsmurfno problem with that
18:10.47b11dok.. so I was the only one on the bus cracking beers at 7:45am..  
18:10.51RoyKaKatty: http://karlsbakk.net/fun/cat.mpg
18:10.51b11dthis is supposed to be minnesota..
18:10.57hbsmurf7:45am?
18:11.05b11dyeah we had to leave early to make it to the game
18:11.08hbsmurfIt's noon somewhere, that's what my grandpa used to say
18:11.19b11dyou cant show up at a baseball game all clear eyed
18:11.19b11d:)
18:11.21ChkDigitIf anyone is in Regina, Saskatchewan over the holidays, The Bushwakker's Blackberry Mead is a must try!
18:11.29b11dRegina fucking rules
18:11.36b11dnext to Saskatoon that is
18:11.36hbsmurfRegina is a bit too far north for me at the moment
18:11.58ChkDigitMeh, I think Saskatoon sucks.
18:12.06rob0b11d: the big deal is/was the acts of perversion you were performing. I mean, I've been drunk, but I never did the kind of things you did with that baseball bat.
18:12.10b11dI was up there for the midwest shriner conference this summer..
18:12.11b11dit was a blast
18:12.19b11dgot put under the table by these 70 year old men drinking scotch..
18:12.20hbsmurfdid they have the little cars?
18:12.24b11dyep.. I drive one
18:12.26hbsmurfAnd the hats?
18:12.27b11dyep
18:12.27hbsmurfthat owns!
18:12.29*** part/#asterisk dmdnb9s (n=dmdnb9s@c-24-20-35-49.hsd1.mn.comcast.net)
18:12.32b11dyeah, they are a blast!
18:12.37b11dthe Fez is well known :)
18:12.43hbsmurfROFL
18:13.14b11dI've got a great photo of me in a little car around here somewhere
18:13.22b11dthey are a bitch to get in and out of though
18:13.28hbsmurfrofl
18:13.33b11dplus its a four-night-a-week commitment to train..
18:13.37b11dwhich sucks
18:13.39SomeOne1SIP/2.0 401 Unauthorized^M
18:13.40SomeOne1wtf?
18:13.41hbsmurfwow
18:13.58SupaplexSomeOne1: sip haxor! ;)
18:14.02hbsmurfI'd say you're unauthorized
18:14.21RoyKaSomeOne1: that's usual, first REGISTER or whatever, then 401 then REGISTER with the correct auth, then 200 or, in case of bad auth, 407
18:14.26SomeOne1dude its totally authorised
18:14.34SomeOne1ahhh
18:14.35hbsmurfAre you sure it's authorized?
18:14.36SomeOne1i see
18:14.38hbsmurfMaybe you THINK it is
18:14.43hbsmurfIt's a feature
18:14.45SomeOne1RoyKa: yeah, i think thats whats happening
18:14.57SomeOne1because later it REGISTERS again
18:15.12RoyKaSomeOne1: as I said, that's according to the RFC
18:15.24SomeOne1and then i get SIP/2.0 100 Trying^M
18:15.28*** join/#asterisk henrique (n=henrique@201-27-71-35.dsl.telesp.net.br)
18:15.31SomeOne1and THEN
18:15.31SomeOne1SIP/2.0 200 OK^M
18:15.31RoyKaSomeOne1: RTFRFC
18:15.39SomeOne1kewl
18:15.42RoyKa:)
18:15.43SomeOne1i was like, what the heck
18:15.44SomeOne1:)
18:15.47hbsmurfThe RFC is confusing, does someone have the Cliff's Notes?
18:16.06hbsmurfThat would work
18:16.15hbsmurfProbably too dry though for mainstream consumption
18:16.21hbsmurfTHE FEEL GOOD MOVIE OF THE YEAR!
18:16.24hbsmurfSIP 2.0!
18:16.26hbsmurfRated R
18:16.31b11dhahah i'd like to see RFC1925: "The Movie"
18:16.38RoyKathe rfc is quite understandable in the way of how client/server chats
18:16.45Supaplexsip cliff noes "talk is cheap. sip admins aren't."
18:16.51*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
18:17.01hbsmurfI like anything that makes me expensive, my wife tells me I'm cheap all the time
18:17.05hbsmurfAnd easy
18:17.15SomeOne1dude it totally like authorizes fine, but when i try to make calls after that
18:17.16hbsmurfOf course, I wouldn't consider myself a sip admin
18:17.21SomeOne1it keeps reading
18:17.34SomeOne1<-- SIP read from 74.52.15.138:5060:
18:17.34SomeOne1SIP/2.0 100 Trying^M
18:17.42hbsmurfanyone here use polycom phones with asterisk?
18:17.50SomeOne1but my asterisk keeps retransmitting the invite
18:17.51SomeOne1Dec 15 13:54:35 VERBOSE[28029] logger.c: Retransmitting #5 (NAT) to 74.52.15.138:5060:
18:17.51SomeOne1INVITE sip:5712417594@carriers.icall.net SIP/2.0^M
18:18.14SomeOne1after retransmit #6 it dies
18:18.20*** part/#asterisk hardwire` (n=hardwire@rdbck-4891.wasilla.mtaonline.net)
18:18.26*** join/#asterisk airwolf__ (n=airwolf@84.241.200.238)
18:18.33SomeOne1any ideas?
18:18.34b11dAirwolf was the best show ever
18:18.38hbsmurfIt was
18:18.39hbsmurfI agree
18:18.44b11dthose black helmets with the little visors..
18:18.46b11dthey rocked
18:18.51SomeOne1knight rider was
18:18.54b11dthe first episode of Airwolf is on google video..
18:19.20b11dknight rider didnt have a ridiculous Earnest Borgnine
18:19.22*** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca)
18:19.28hbsmurfKnight Rider wasn't believeable
18:19.31hbsmurfAirwolf was
18:19.34b11dAirwolf was totally believable
18:19.35b11d:)
18:19.38robl^no!  the best show ever was The Great Space Coaster.
18:19.41mercestesThe A Team was the best show ever.
18:19.41b11dI thought it was a documentary actually
18:19.43*** join/#asterisk zotz (n=zotz@24.244.163.157)
18:19.48hbsmurfrofl
18:19.50b11dA Team was great..
18:19.55mercestes:D
18:20.03blitzrageI have Season 1 on DVD :)
18:20.05b11dnice!
18:20.11hbsmurfI love watching the A Team reruns on the Sleuth channel or whatever it is
18:20.15Kattylters_: that's too much work.
18:20.19Kattylters_: don't be riddicurus.
18:20.24b11danyone ever watch EMERGENCY!
18:20.25b11d?
18:20.31b11dwith John Gage and Roy Desoto
18:20.38hbsmurfYes
18:20.40hbsmurfI seem to remember that
18:20.50SomeOne1does anyone know where i can find this SIP authehtatication procedure
18:20.50b11dhahah with doctor Brackett and Joe Early
18:20.52SomeOne1and stuff
18:20.54SomeOne1like the codes
18:20.57SomeOne1such as 401
18:20.59SomeOne1200
18:20.59SomeOne1100
18:21.05b11dRFC for SIP i'd assume
18:21.07b11dwww.faqs.org
18:21.13*** join/#asterisk toweliee (n=phph@do.you.like.my.frippers.com)
18:21.22b11dcheck this out
18:21.23b11dhttp://www.iptel.org/book/print/6
18:21.25SomeOne1http://www.faqs.org/rfcs/rfc3261.html
18:21.25b11dit might have it
18:21.31SomeOne1found it
18:21.33SomeOne1thanks
18:21.34b11dgood SIP intro someone posted here yesterday
18:21.36b11dcool.. np
18:21.52b11dcheck out that other link too though, if you're learning about SIP
18:22.09*** join/#asterisk bkruse (i=bkruse@nat/digium/x-c61864d7350781ed)
18:22.22mercestesI love you katty!
18:22.35hbsmurfInteresting, I need to find time to read that
18:22.42bkrusemercestes: please take your chat to the #<3 room....
18:22.43bkruse:P
18:22.47b11dyou run your own life..  make the time..
18:22.48b11d:)
18:22.49mercesteslol
18:22.53b11dhaha
18:22.57resistancewhat is the best way to upgrade to the latest v of trixbox
18:23.16mercestesresistance:  follow the directions they give you in #freepbx when you ask there.
18:23.19b11dask in #freepbx
18:23.19b11dyeah
18:23.20towelieehiya all, i need some help with using call-limit in sip.conf, mainly i need a way to select which sound file plays back when the user cannot make more calls, is it possible?
18:23.29resistanceok, thanks
18:23.34b11donly if you smoke your last joint with me toweliee
18:23.41hbsmurfI've got a wife and kids and a computer business, I have no time to myself
18:23.42hbsmurf:)
18:23.43towelieeLOL :P
18:23.45rob0resistance is futile
18:23.56b11dthats the general excuse I hear these days :)
18:23.58bkrusetoweliee: sounds like dialplan logic more than anything, does asterisk set a variable?
18:24.12mercestesyour ass will be laminated.
18:24.13Supaplexresistance is V/(I*R)
18:24.15bkruseyou could say exten => whatever,1,Gotoif(${CALLIMIT bla blah
18:24.20mercestes...or something like that.
18:24.21hbsmurfI've got a list of things I'm going to try to learn this weekend starting with Realtime and DUNDi.  I hope I get somewhere.  :)
18:24.22b11dasslamination..  I like that..
18:24.30hbsmurfThat might hurt
18:24.33hbsmurfWHat if you have to go?
18:24.42bkrusehbsmurf: dundi will hurt, but will be awesome when you set it up
18:24.49resistancetry photocopying u're ass and falling through
18:24.57towelieebkruse: thanks maybe I'll try that, by default it plays the "your call cannot be completed as dialed", which is not the appropriate thing to play in such a situation
18:24.58resistancemost office accidents happen that way
18:24.58*** join/#asterisk qwertz (n=qwertz@82.193.233.238)
18:25.16bkrusetoweliee: agreed, asterisk has to set some kind of variable, just not suer what it is
18:25.27mercestesresistance:  LMAO!!!  OMG, and then the hot incandesent bulb starts hitting you...muhahaha
18:25.37towelieebkruse: k i will do some more research
18:25.45hbsmurfDigium support has such nice music on hold!
18:25.55caio1982AC/DC tunes?
18:26.03hbsmurfNah, some classical thing
18:26.10bkrusetoweliee: is it the incoming call limit?
18:26.13mercestestoweliee:  try throwing a "goto s-${DIALSTATUS}" and see where it tries to dump you.  maybe that will help
18:26.14b11dI like cxtec's automated system..  ever call them?
18:26.16resistancetalk about roasting u're balls
18:26.30b11d"if you've ever had to pull a cable, press 1.. if you know what SIP means, press 2" etc
18:26.32bkrusetoweliee: ahh! i found it
18:26.33mercestesChestnuts roasting on an open fire.....
18:26.38qwertzHi, trying to install * bristuff version 0.3 on debian sarge but when compiling the gsm pci driver I get the error "Makefile:266: /usr/src/linux-headers-2.6.17-2/scripts/Kbuild.include: No such file or directory" - so could any debian user tell me where this file is located in debian or how to get it?
18:26.38bkrusemercestes: good idea
18:26.40bkrusetoweliee: http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Limit+Exit+Options
18:26.44hbsmurfSIP?  Isn't that what I do when drinking my coffee in the morning?
18:26.47b11d:)
18:27.00mercestesqwertz:  Did you try #debian?
18:27.01bkruseqwertz: you have kernel headers installed?
18:27.16bkrusethe build needs variables from the kernel makefile
18:27.21b11dI understand Quantas has the best automated system in the world.. but i've never called it
18:27.27towelieebkruse: yup found it too ${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits and ${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
18:27.51Supaplexqwertz: install apt-file, apt-file search <file|path>
18:27.52bkrusetoweliee: awesome! glad to be of assistance :D
18:27.53bkrusekinda
18:27.57towelieethanks heh :P
18:28.14bkruseqwertz: apt-get source bristuff
18:28.28bkruseqwertz: or even better, download the misdn packages from digium
18:28.47mercestesor buy a cisco call manager.
18:28.48b11dmy account rep at Cxtec just sent me a photo of himself in drag..
18:28.51b11dthats disturbing as hell
18:28.59caio1982qwertz: run 'm-a prepare' (from the module-assistant package) and then retry
18:29.14mercestesb11d:  Post it on th einternet.
18:29.18hbsmurfI'm looking forward to DUNDi.  
18:29.25hbsmurfIt should be interesting
18:29.26b11dits from their halloween party.. thats a relief..
18:29.50Supaplexhaha
18:29.54mercestesbkruse:  #<3 is boring.
18:29.58caio1982qwertz: anyway, debian provides a bristuffed version of asterisk already
18:30.04bkrusemercestes: hahahahaha
18:30.07caio1982and there are backports of it for sarge
18:30.16b11dI love you guys
18:30.18Supaplexand they work great!
18:30.29mercestesI love you too, b11d.
18:30.33*** part/#asterisk bkruse (i=bkruse@nat/digium/x-c61864d7350781ed)
18:30.37hbsmurfYou can't have my beer!
18:30.37hbsmurfno!
18:30.39b11dthanks :)
18:30.41b11dhaha
18:30.49*** join/#asterisk bkruse (i=bkruse@nat/digium/x-c61864d7350781ed)
18:31.26caio1982bkruse: wasnt you the guy that joined debian-pkg-voip and sais that got some alpha .debs of 1.4?
18:31.29Supaplexback so soon bkruse?
18:31.30caio1982said*
18:31.41bkrusecaio1982: yes
18:31.48hbsmurfHow difficult is realtime?  I found a whitepaper from Astricon and it talked about realtime and dundi.  Is it a pain in the rear?
18:31.49bkrusecaio1982: there almost done, ill get time to work on them tonight
18:31.52caio1982bkruse: can I take a look at their diffs/sources?
18:31.54bkrusewell 1.4 beta3
18:32.05qwertzthanks for all the hints, didn't know that there is an bristuff version as debian package so I'll try this at first
18:32.06mercesteshbsmurf:  It is so painfully simple that it is unnecessarily difficult.
18:32.06bkrusecaio1982: ya ill send you the debianized tarball and the diff's
18:32.13hbsmurfcool
18:32.13b11dwell, when I go and rip the power cable from my primary asterisk PBX, I get less than 1 second of "down time" .. not bad..
18:32.16hbsmurfcoolcoolcool!
18:32.29mercesteshbsmurf:  If they had actually required some worthwhile steps to make it work, some that actaully requried documentation...something someone would write a WIKI on....then it would be easier..
18:32.32b11dlooks like Airwolf's gone stealth..
18:32.36caio1982hbsmurf: the major problem is that there is no "real" realtime yet... try the static schema and you're done
18:32.46mercesteshbsmurf:  But since it's just kinda ...'there' ...it's nearly impossible.
18:32.48bkrusecaio1982: im also getting zap 1.4 and misdn for debian
18:32.48hbsmurfI've got three boxes at home I'm going to work on this weekend.  I want to learn how to load balance stuff
18:32.49bkruseBUT
18:32.53caio1982bkruse: awesome! (although would be nice to have them online)
18:32.56bkruseyou have to build it for so many kernels :X
18:33.13bkrusecaio1982: they will be in debians experimental repo and then pushed to ubuntu's repo ( if theyll accept it )
18:33.25bkrusemercestes: these kids in #<3 are wierd.
18:34.01hbsmurfmercestes:  I'm intrigued by your ideas and would like to subscribe to your newsletter.
18:34.04caio1982bkruse: i know, i track the list threads :) but i believe some peer review will be necessary before that, and also i think that pkg-voip people will prefer to change some stuff, probbably :)
18:34.12hbsmurfI'll read the WIKI, I promose
18:34.14hbsmurfpromise too
18:34.17hbsmurfdamn fingers!
18:34.17mercesteshbsmurf:  If you want load balancing, check out #ser.
18:34.50hbsmurfI've read a bit about ser
18:35.13hbsmurfI'm only talking 300 clients
18:35.16hbsmurfdo I really need ser?
18:35.51wasimyes ser, yes ser, three bags full ...
18:35.57hbsmurfcool
18:36.01hbsmurfI guess I'll read more then
18:36.04silget an SBC
18:36.13hbsmurfSouthwestern Bell Corp?
18:36.27hbsmurfSouthern Baptist Convention?
18:36.40silsession border controller
18:36.43hbsmurf:)
18:36.47b11dSick Balls Corp.
18:36.47hbsmurfI just found that in google
18:36.54hbsmurfor slick balls corp!
18:37.01caio1982bkruse: how can i get the stuff? 8)
18:37.16silhttp://infiltrated.net/mydesk/nCite.jpg
18:38.46hbsmurfHey, those wires are way too organized!
18:39.10b11dyeah.. must be photoshopped
18:39.10b11d:)
18:39.52hbsmurf:)
18:40.26*** join/#asterisk Primer (n=vi@sh.nu)
18:40.39bkrusecaio1982: be on tonight when i run my build
18:41.00b11dGET BACK IN <3 GOD DAMMIT
18:41.14RoyKasterisk 0.3.0?
18:41.15RoyK:D
18:41.16PrimerTrying to determine what the status numbers returned by a QueueMember event in the manager console mean...can't seem to find any info on it since "status" is such a common word...thoughts?
18:41.35bkrusecaio1982: im REALLY good friends with the pkg-voip guys, they are amazing, and im just starting a new experimental part with beta3
18:41.42bkrusethey currently have beta2, that doesnt wrok
18:42.33caio1982bkruse: unfortunately i'm leaving for a vacation time tonight ;-) because that i was prefering an URL for later checkings, but that's okay
18:44.05seele_how can i announce a call before transfer it?
18:44.21hbsmurfAnnounce how?
18:44.27hbsmurfTo the person answering the phone?
18:44.56seele_hbsmurf, call a person ... announce the call .. and tranfer the call to the person
18:45.08hbsmurfThat should be pretty straightforward
18:45.19hbsmurfThe person answers the call and you use Playback() to announce
18:45.23hbsmurfI would think
18:45.33hbsmurfand after the announcement you connect them?
18:45.43hbsmurfwait
18:45.57hbsmurfMaybe hangup and then dial them again with the real call
18:46.01hbsmurfthat is probably the hard way though
18:46.42hbsmurfI'm still working on my elegant coding so take whatever I say with a grain of salt
18:46.47*** part/#asterisk henrique (n=henrique@201-27-71-35.dsl.telesp.net.br)
18:47.35*** join/#asterisk dasenjo (n=dasenjo@208.195.215.71)
18:48.18b11d<3 is a great channel
18:49.55robl^b11d: only on odd numbered days ;-)
18:50.01b11dhahaha
18:50.10b11dwhat side of the date line are you on? :P
18:50.52robl^hehe..  today is an odd day..  for about 11 more hours
18:51.02b11dyou are CST like me then
18:51.22*** join/#asterisk s1gny|wrk (n=s1gny@p54915480.dip.t-dialin.net)
18:51.33*** part/#asterisk s1gny|wrk (n=s1gny@p54915480.dip.t-dialin.net)
18:51.42*** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
18:51.49*** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
18:52.10hbsmurfWow
18:52.18b11dso many people use real names here
18:52.44robl^my real name isn't pronouceable by humans. ;-)
18:52.52hbsmurfabe went from 120 calls in version A to 40 in version B.  Wow.
18:52.54b11dI always knew you were a reptilian..
18:52.58b11dDavid Icke was right all along
18:53.07*** join/#asterisk andresmujica (n=AndresMu@201.245.236.158)
18:53.10robl^hehe
18:53.22robl^abe?
18:53.28b11dAsterisk Business Edition
18:53.33robl^ohhh.
18:53.33b11dI learned that earlier too
18:53.59hmmhesaysi learned a kenny chesney tune last night
18:54.01hmmhesayschicks love that
18:54.05robl^ppl abrv 2 much
18:54.08hbsmurfCounty music chicks are hot
18:54.08b11dhaha
18:54.16b11dyeah they can be..  asian rave girls are hotter
18:54.27hmmhesayscountry chicks do anal
18:54.36b11dwhat chicks dont?
18:54.36*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
18:54.36hmmhesayscause they're from trailer parks n shit
18:54.51robl^teletubbies in leather are even hotter
18:54.57b11dfuck thats hot
18:54.58ShadowHntri won't ask.
18:54.58b11ddont touch me
18:55.14hmmhesaysmy girlfriend doesn't
18:55.30b11dthen you just need to start..
18:55.34hbsmurfAll girls love anal, they just don't know it yet
18:55.37b11dshe'll love it..
18:55.41b11dhbsmurf is correct
18:55.48hmmhesayswell if she wouldn't slap it away every time I try to creep in
18:55.49b11dalso.. there you need 'tact' -- at least, at first..
18:55.54*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
18:56.01*** join/#asterisk xnon (i=xnon@200.8.5.123)
18:56.54b11dI dislike the slew of xmas cards that arrive in my mailbox from salesmen this time of year.
18:57.10b11dI should write back "im jehovas witness you son of a bitch"
18:57.17hmmhesaysyou'd get less if you shot everyone that sent you one
18:58.00hbsmurfYou need to get her in the right mood
18:58.03hbsmurfI suggest alcohol
18:58.12hbsmurfOnce you get her to do that she'll be begging for it
18:58.18hmmhesaysdude you dont' need to school me
18:58.23hbsmurf:)
18:58.29b11dhmmhesays can handle himself
18:58.30hbsmurfNo schooling, encouragement!
18:58.37b11dyeah.. anal rules
18:58.42hbsmurfLet's not talk about handling oneselves
18:58.43hbsmurf:)
18:58.53b11dnow lets get into the ass-to-mouth discussion as seen in Clerks 2
18:59.02hmmhesaysb11d: i'm going to add a (if you're on on bottom) disclaimer
18:59.10b11dlol
18:59.12hmmhesays*not on bottom
18:59.46b11danyone ever get the "dead log" of a woman?
18:59.49b11dthats the worst..
18:59.55hbsmurfThey just lay there?
18:59.56b11dyes
19:00.01hbsmurfI'm not seeing what's wrong with that
19:00.03hbsmurfI'm not there for her
19:00.04hbsmurf:)
19:00.06b11dHAHA
19:00.09b11dthat rocks :)
19:00.19b11dim sending you $20
19:00.44hbsmurfheh
19:00.49hmmhesayswow
19:02.29b11di'd definatly have to say im more of an assman as I get older..
19:02.43irqi am definitely an assman and have always been an assman
19:02.46b11d(btw, the filler discussions in here never get old)
19:02.54b11dthats respectable irq..
19:03.09hbsmurfI love how I can ask about realtime and talk about anal in this channel
19:03.10hbsmurfthis rocks!
19:03.12b11d:)
19:03.18irqmy pride and joy: http://zeppelin.stepahead.net/~dan/upsgirl/
19:03.42b11dthose are nice!
19:04.00irqalso http://zeppelin.stepahead.net/~dan/pz/
19:04.31b11dim making that the homepage for all browsers on campus
19:04.42irqwhich one?
19:05.38b11dchrist thats a nice ass :P
19:05.38b11d<PROTECTED>
19:05.38irqheh yeah, that's why i kept those sets... they are the best asses ever
19:05.39irqb11d: cool, but would you mind mirroring the files to a campus-local machine? i don't want my home cablemodem raped...
19:05.39b11dnp..  i'll map a drive..
19:05.45irqimage22 is particularly good
19:06.06b11dsigh.. why.. im at work..  
19:06.07b11dugh..
19:06.08irqanyway, this doesn't really help the asterisk scene very much :)
19:06.13irqheh i'm at work too, also i have a 30" lcd
19:06.15b11dno ones talking about asterisk anyway
19:06.19irqwhich really, really helps the images
19:06.22hmmhesaysbah I remember that chick from fusker
19:06.22b11d30 eh.. wow..  22" here
19:06.25irqbut i'm in an office and no one can see my screen
19:06.32irqi bought it with my own money, company wouldn't budge
19:06.36irqyou can get them from dell now for $1100
19:06.42b11dyeah thats how it goes :)
19:06.59hmmhesaysirq, where'd you get those i've seen them floating around the web for years
19:07.08hmmhesays22 you're just a pup
19:07.10irqjust keeping my eyes open
19:07.20irq22? what?
19:07.22b11dits not the size that matters
19:07.23*** join/#asterisk rr-- (n=rr@cpe-66-69-217-206.austin.res.rr.com)
19:07.26b11dits how you use the display space..
19:07.30b11dits THE RESOLUTION DAMMIT
19:07.30irq4chan.org's /s helps
19:07.35hmmhesaysdid you just say you were 22
19:07.38irqthe ups girl images i got from a r/c helicopter friend
19:07.41irqno he said 22"
19:07.45irqas in that's the size of his LCD
19:07.46hmmhesaysahaha
19:07.51b11dim only 25..
19:08.01b11d"only" -- I feel so old..
19:08.19irqi blew up the engine in my car yesterday :(
19:08.20b11dits the system admin life.. it'll beat the hell out of you..
19:08.25b11dthat sucks...
19:08.31b11ddid you impress any girls by doing it?
19:08.40hmmhesaysyes recycled net images
19:08.48hmmhesaysi want to see some originals here
19:09.01b11dI'll show you mine.. hell. it'll be your nightmare, not mine.
19:09.07hmmhesaysb11d: thats why I hung up my sys admin coat and went into contract work
19:09.10irqb11d: no. it just pissed off my wife
19:09.23b11dno doubt :)
19:09.24hmmhesaysnow I play guitar, bring the rock, and do some work
19:09.28irqi've been a sysadmin for only about 6 years now but so far i love it
19:09.29b11dhmmhesays.. I like the sound of that..
19:09.34irqhey i bring my guitar to work too!
19:09.40silcontract work blows ... i did contract work for IBM from home
19:09.44sili got bored out of my mind
19:09.55hmmhesaysi don't get bored
19:10.02b11dyou only get bored if you're boring..
19:10.04hmmhesaysI have plenty of computer games, and a guitar
19:10.05irqif i get bored, i play my ds lite and hide under my desk
19:10.21b11di'd lay back and play the intro to Computer Man over and over..
19:10.30hmmhesayslol
19:10.35hmmhesaysso did you make it to the music shop?
19:10.36*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
19:10.37sili have a 5yo @ home ... during the summer (while i was working) all i heard was "play with me play with me play with me"
19:10.41b11dI dont want to talk about it
19:10.42b11d:)
19:10.46b11dno.. I didnt :(
19:10.49hmmhesaysget down there damnit
19:10.52silno benefits... a lot of money ... but not worth it
19:10.59b11di'll go tomorrow afternoon..
19:11.00qwertzIs there some info available what the values in the top line (holdtime, A, C, SL) of the show queues command are?
19:11.11b11dsil..  thats fucked :)
19:11.20hmmhesayssil: doing your taxes kind of sucks
19:11.28hmmhesaysfilling out the estimated tax forms
19:11.41silyea it does ... but on the flip side i wrote off a lot of stuff
19:11.51hmmhesaysyeah you can write off almost everything
19:12.10hmmhesaysrent, gas
19:12.19*** join/#asterisk blackgecko (n=blackgec@189.142.42.162)
19:12.24hmmhesaysoffice supply's
19:12.24silits good when youre young or in school (i'm 33) but not good when you have a family (benefits)
19:12.31silduring the dotcom days i used to love it
19:12.31hmmhesayssil: yeah
19:12.37silheck who didnt ;)
19:12.39hmmhesayssil: i'm 24 with no family
19:12.57silmy coworkers here are almost all under 26
19:13.02blackgeckoanyone sucesfully instaled asterisk on ubuntu ??
19:13.04silthey love it all bank
19:13.25hmmhesaysblackgecko: you should be able to get it to run on that bastardized copy of debian
19:13.27b11d25..  no family..
19:13.39b11ddrug user..   ex-alcoholic..
19:13.46b11d:P
19:13.48b11d..fucked..
19:13.49*** join/#asterisk _spirit_ (n=spirit@66.161.100.230)
19:14.00hmmhesaysalcoholic, did a few back in the day
19:14.07*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)
19:14.07blackgeckohmmhesays: its a friens lap not mine
19:14.07shellsharkb11d: that's you?
19:14.12b11dits me
19:14.14silheh cant be too bad my boy works for (dare i say it) MS out of amsterdam... ive never seen him sober
19:14.15hmmhesayshell 90% of the population of fargo is an alcoholic
19:14.23silhe's like 27 now and he enjoys it
19:14.25*** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com)
19:14.26blackgeckohmmhesays: but im unable to make compile
19:14.33hmmhesaysmake compile?
19:14.35b11dI'd love to work in amsterdam
19:14.36shellsharkb11d: i've been sober 1 year as of december 1st :)
19:14.39hmmhesaystry just "make"
19:14.42b11dcongrats man :)
19:14.48b11dfeels good doesnt it
19:15.28shellsharkb11d: longest i've gone since before i started drinking ;)
19:15.28hmmhesaysmake sure you sudo -s first
19:15.28shellsharkb11d: which was ~13 or so
19:15.29b11dthats about the same time I started into "it"
19:15.29hmmhesaysi didn't start drinking till i was 19
19:15.29irqi sometimes bring a flask to work
19:15.29blackgeckohmmhesays: yeah im doing just make but get stuck at chan_phone
19:15.29b11dbut being from Ontario.. its not as strange..
19:15.29irqof scotch, lagavulin 16 specifically
19:15.32shellsharkyeah it feels good... makes me want to go out and grab a pint in celebration...
19:15.32hmmhesayspm me
19:15.41shellsharkthen i'm all like "oh yeah" and it sucks ;)
19:15.43b11donce you learn to drink scotch properly, its a whole new world..
19:15.49b11dhahahah shellshark..
19:15.54hmmhesaysvodka is my poison
19:15.55blackgeckohmmhesays: chan_phone.c:41:29: error: linux/compiler.h:
19:15.58b11ddont do it..  if you cant handle it.. dont do it :)
19:16.06b11dI should send you some "Silent Sam" from Canada hmmhesays..
19:16.07hmmhesaysblack gecko, comment that out of chan_phone.c
19:16.09b11dyou cant taste or smell it
19:16.10hmmhesaysits not needed anymore
19:16.27hmmhesaysb11d: properly chilled shakers vodka is the same way
19:16.32b11dshakers eh..
19:16.33b11dnever had that
19:16.33*** join/#asterisk QbY_ (n=Kelvin@cm-64-221-168-170.dhcp.southerncoastalcable.net)
19:16.41hmmhesaysmade in our home state
19:16.47b11dcool..
19:16.51hmmhesaysbenson MN
19:16.53sili used to hit vodka straight up ... ;) absolut and grey goose
19:17.05b11dabsoult tastes like rubbing alcohol smells.
19:17.06hmmhesaysabsolut is shit compared to shaers
19:17.12hmmhesays*shakers even
19:17.18irqhere's those previously mentioned pics, but all on one html page so you don't have to click around: http://zeppelin.stepahead.net/~dan/pz/list.html
19:17.21monstedzubrowka is good
19:17.26silhmhesays: i lived in sweden for a little while ;)
19:17.27b11di need to experience this "shakers" -- but as of now, Silent Sam is #1 to me
19:17.37silobviously its going to be the vodka of choice over there
19:17.41hmmhesaysyeah
19:17.44hmmhesaysvox is good
19:17.53b11dthank irq.. you bastard
19:17.58*** join/#asterisk QbY (n=Kelvin@cm-64-221-168-170.dhcp.southerncoastalcable.net)
19:18.07blackgeckohmmhesays: what part should i comment ?
19:18.30hmmhesays#include linux/compiler.h
19:18.49_spirit_I just got finished talking to my boss about asterisk, were looking for a consultant. Someone with experience with asterisk in a call center environment, and setting up/choosing the T1 hardware/lines. AKA who wants to make $$ =) Southern California Orange County area.
19:18.54hbsmurfI love chocolate filled thank you gifts this time of year!
19:18.55hmmhesaysor remove it completely
19:18.55b11dhttp://zeppelin.stepahead.net/~dan/pz/Image50.jpg
19:18.58b11dthats the one
19:19.17hmmhesaysirq: i saw all those on fusker on the same page a few years bac
19:19.20hmmhesays*bac
19:19.24hmmhesaysbah fucking keyboard
19:19.25hbsmurfAh, that brazillian girl
19:19.28hbsmurfshe's nice
19:19.39irqb11d: heh, i just remembered, when i first found those pictures, it was while i was interviewing someone for a job
19:19.45shellshark_spirit_: do they have to be local? i wouldnt mind flying out to do it
19:19.46b11dLOL
19:19.49hbsmurfYou surf porn while interviewing?
19:19.51shellshark_spirit_: i'm in Illinois
19:19.51irqb11d: it was really difficult concentrating
19:19.56hbsmurfDude
19:19.56b11dundoubtely
19:19.59hbsmurfYou hiring?
19:19.59shellsharkhbsmurf: why not?
19:20.00irqhbsmurf: no, i don't, but a friend IM'd me the link and i just clicked on it
19:20.05hbsmurfrofl
19:20.06irqhbsmurf: and once i was there, wlel, i couldn't just go away
19:20.15hbsmurfNevermind, I'm the boss here so I can surf
19:20.16irqsince then i don't even bring my laptop to interviews
19:20.19blackgeckohmmhesays: damn ive removed and still error
19:20.28hmmhesayshuh?
19:20.38hmmhesaysyou can't be getting that error if you removed that line
19:20.38_spirit_My company isnt big enough to pay to fly some one out and have them stay here... it will prob be a one or two month job
19:21.01shellsharkso you're not going to pay $$ then, you're just going to pay $?
19:21.14hbsmurf$ is less than $$
19:21.14blackgecko#if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,0)
19:21.14blackgecko# include <linux/compiler.h>
19:21.14blackgecko#endif
19:21.15blackgecko#include <linux/ixjuser.h>
19:21.18shellsharkugh, false advertisement...
19:21.19b11dugh..
19:21.19b11dwow..
19:21.20b11dhttp://www.npr.org/templates/story/story.php?storyId=6626823
19:21.22b11dthats pathetic
19:21.28rr--if some fxo or fxs ports are on sangoma A200 cards and other ports are on a digium TDM400P and X100P cards ... this won't be a problem for asterisk to handle, will it? asterisk can handle multiple PCI cards made by different manufacturers, right
19:21.30blackgeckothats what ive removed
19:21.33hbsmurffigures
19:21.46hbsmurfAsterisk can handle multiple cards
19:21.50b11dthe Cowboys are building a new stadium worth $1 BILLION?
19:21.52b11doh thats NICE.
19:21.54hmmhesayshmmm maybe you should put that line back in
19:21.56b11dwhat a good use of money..
19:21.58b11dfuck..
19:22.05skachaha.
19:22.06hbsmurfMy God
19:22.15hbsmurfA 240 call license for abe is now $3315 cost
19:22.16hbsmurfwow
19:22.17skaci prefer being called jesus.
19:22.21hbsmurfCisco Call Manager is that much
19:22.27shellsharkb11d: yeah man, and think of all the homeless people in dallas, and the kids with no food...
19:22.44shellsharkhbsmurf: abe?
19:22.45b11dexactly!!
19:22.50hbsmurfasterisk business edition
19:22.53rwxr--r--shellshark ... what do you mean the cowgirls' needs supercedes human necessities
19:23.03rwxr--r--obviously you dont have the mind of a politican :X
19:23.06b11d:)
19:23.07hbsmurfI'm running 1.2.13 here at the office and I run business edition for my clients
19:23.14shellsharkrwxr--r--: build it and they will "come" right?
19:23.18b11dlol
19:23.21hmmhesaysblackgecko: pm me
19:23.39shellsharkhbsmurf: oh, asterisk business edition...
19:23.44hbsmurfyeah
19:23.46rwxr--r--yup ... see theyre looking out for the people ... the homeless people can bunk under the crosswalks of the new stadium
19:23.53rwxr--r--and hunger? ...
19:23.58hbsmurfAnyone know of a way to send messages to Polycom phones?
19:24.04hbsmurfMake something display on them?
19:24.06rwxr--r--well not a disposed hotdog would go to waste
19:24.43b11dlets not put that money into education..
19:24.46Qwell[]rwxr--r--: You forgot the - at the start
19:24.47b11dthat'd be a bigger waste
19:25.05rwxr--r--education ... its covered too
19:25.14shellsharkb11d: edumatcaion? hoo nedz dat?
19:25.18rwxr--r--1st down ... 3 more downs to go (subtraction)
19:25.20linageeis sellvoip or voipstreet more reliable than voicepulse?
19:25.32rwxr--r--field goal = addition
19:25.37shellsharklinagee: shellshark.net is more reliable then voicepulse :)
19:25.49hbsmurfshellshark:  How do you really feel?
19:25.59shellsharkhbsmurf: about?
19:25.59*** join/#asterisk ManxPower (n=manxpowe@231.sub-75-201-47.myvzw.com)
19:25.59linageeshellshark: interesting how <voipname> = <your_nickname> hehe
19:26.02nays85linagee : imho, no way
19:26.14hbsmurfshellshark:  voicepulse  :)
19:26.19shellsharklinagee: it's a great way to advertise :)
19:26.21linageeshellshark: i want to port my cell phone number to a voip number
19:26.29shellsharklinagee: no problem
19:26.43nays85linagee : they all say "no problem", he won't be able to do it
19:26.47linageelol
19:26.56shellsharknays85: why not?
19:27.03shellsharknays85: WNP is your friend
19:27.07*** join/#asterisk Dr-Linux|home (n=Dreamer@202.59.73.131)
19:27.19nays85because, like i said, they all say "no problem" and a month later say they can't do it
19:27.25linageeshellshark: also, i can only do IAX because of my firewall
19:27.28b11dmamma talkin to me tryin to tell me how to live..
19:27.37shellsharklinagee: no problem
19:27.45rwxr--r--so in a few months... i will have this IDS for asterisk done :X
19:27.58hmmhesaysIDS?
19:28.24shellsharklinagee: we offer IAX2, SIP, H323, MGCP, SCCP, STIM, and whatever various things unicall supports :)
19:28.24rwxr--r--intrusion detection system
19:28.43Qwell[]shellshark: better be careful when you say SCCP...  it's quite ambiguous, depending on the context
19:28.49rwxr--r--im calling it RAPID (Robust Asterisk PBX Intrusion Detection) ;)
19:29.01nays85linagee : they can offer you anything you want, but they can't pay for an ssl cert before pimping their website in #asterisk :b
19:29.05*** join/#asterisk stealthmethod (n=123@70.46.114.23)
19:29.07shellsharkQwell[]: true
19:29.12Qwell[]rwxr--r--: RAID would be way cooler
19:29.12hbsmurfCan we get some decent skinny support for the 7920?  Who do I have to buy pizza and beer for to get that?
19:29.20Qwell[]hbsmurf: That would be me
19:29.29Qwell[]7920 works great in chan_skinyn in 1.4
19:29.29rwxr--r--ppl will get confused with disks arrays ;)
19:29.29shellsharknays85: it's coming with the new website
19:29.37hbsmurfGive me an address and how much and I'm there
19:29.37hbsmurf:)
19:29.39shellsharknays85: due in january
19:29.42hbsmurf1.4?
19:29.43rwxr--r--i despise cisco phones :\
19:29.45hbsmurfI'll try that tonight
19:29.52Qwell[]1.4 beta anyhow, heh
19:29.52hbsmurfOk
19:29.55hbsmurfI haven't tried it with 1.4 beta yet
19:29.58hbsmurfjust 1.2.13 and trunk
19:29.58Qwell[]hbsmurf: If it works, feel free to still buy beer and pizza ;)
19:30.01hbsmurf:)
19:30.30nays85shellshark : you and the 10 providers who did the same exact thing in here are all out of business... it's like the same pattern over and over again
19:30.30hbsmurfIf I can get the 7920 working and hooked into an OLD call manager system I'll send you pizza and beer
19:30.31hbsmurf:)
19:30.32shellsharknays85: i'm not out of business ;)
19:30.36*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
19:30.39Qwell[]as long as the CCM supports SIP, it should be fine
19:30.45hbsmurfIt doesn't
19:30.47hbsmurfh.323
19:30.51hbsmurfIt's OLD
19:30.52shellsharknays85: i've been doing this for a year, and have been profitable the entire time, unlike all the graveyard junkies
19:30.52Qwell[]god help you if you have to use mgcp or something...or h.323 :(
19:30.52hbsmurf3.2
19:30.54nays85again, that's what they all say
19:30.57hbsmurf:)
19:31.11De_Monhow would I send someone into a conference? I want to dial a number and have it put me in a conference and then call someone else and put them in the conference with me
19:31.12nays85how many paid full-time employees do you have?
19:31.20shellsharknays85: none :)
19:31.26hbsmurfQwell[]:  Put the system in over 4 years ago and haven't done a damn thing with it since then.  It just runs.  We're going to propose replacing it with Asterisk when it starts dying
19:31.44nays85so who's going to answer support emails and calls?
19:31.51shellsharknays85: that's the key to being profitable, make everyone on-call contractors... it's great
19:31.58hbsmurfOf course, this 40 call limit in the new ABE is going to be a pain.  :)
19:32.18Qwell[]hbsmurf: You can increase the limit
19:32.19hbsmurfshellshark:  I just tell my employees to bill out so much a month or they're gone!
19:32.20hbsmurf:)
19:32.21Qwell[]up to 240 I think it was
19:32.30shellsharkhbsmurf: but of course :)
19:32.33*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
19:32.37nays85linagee : a good rule of thumb around here is to stick with providers that have been around the longest
19:32.52nays85meaning selling service, answering emails, picking up the phone the longest
19:32.53shellsharknays85: you're awful negative
19:33.04nays85not people who say they've been 'in the business' the longest
19:33.15hbsmurfQwell[]:  $3315 cost for 240 licenses.  Ouch!
19:33.16Zodiacalanyone know if theres a good speach-to-text program for *?
19:33.17shellsharknays85: i've got a 95% customer retention rate
19:33.20nays85because one-man operations have screwed a lot of people in here and their customers
19:33.23linageeshellshark: if your business fscks me over, i can always LNP to someone else, right?
19:33.41linageemy number doesn't get locked into your system or something?
19:33.59shellsharklinagee: it's possible to port it somewhere else later, sure
19:34.07linageeinteresting
19:34.10Dr-Linux|homeQwell[]: Sergio?
19:34.15shellsharklinagee: of course you'd have to pay to port it again
19:34.21Qwell[]Dr-Linux|home: might as well be dead
19:34.31linageeshellshark: what is the fee you charge to port? everyone seems to charge an ambigous fee
19:34.32De_Monshellshark thats something the telcos charge isnt it?
19:34.35Dr-Linux|homehhm..
19:34.38nays85shellshark : what's the full name and address that shellshark is incorporated under?
19:34.38Qwell[]hbsmurf: I don't know the pricing..  you'll have to ask Sales
19:34.49Dr-Linux|homeQwell[]: does 4.1 solves my problem?
19:34.55shellsharknays85: it's not incorporated, it's a general partnership
19:34.56Qwell[]Dr-Linux|home: 4.1?
19:35.03De_MonDr-Linux|home 1.4?
19:35.04hbsmurfQwell[]:  I'm looking at netxusa which is where I buy it from.  
19:35.04Dr-Linux|home1.4
19:35.09Dr-Linux|homebeta
19:35.10shellsharklinagee: $25 one time
19:35.11Qwell[]hbsmurf: ahh, okay
19:35.34*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
19:35.36shellsharkDe_Mon: yessir, we outsource LNP, as we don't have SS7 trunks ourselves
19:35.36hbsmurfQwell[]:  It went from 120 calls to 40 calls between A and B which is kinda scary
19:35.46hbsmurfQwell[]:  I'm waiting for a call back from sales
19:35.54rr--is this correct -- an FXS port (or telco line) can be thought of as a 'source' and an FXO port is like a 'sink' ... you can connect a single FXS port (or telco line) to multiple FXO ports, but you cannot connect multiple FXS ports to an FXO port ?
19:36.06Dr-Linux|homeLumenVox looks nice, but if i could have installable demo
19:36.16hbsmurfrr--:  I think you're backwards
19:36.24hbsmurfrr--:  Nevermind
19:36.30ManxPower~fxofxs
19:36.40jbothmm... fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
19:36.40Qwell[]You wouldn't want to connect multiple lines to a single FXO
19:36.40hbsmurfrr--:  YOu're right, I'm thinking of it backwards
19:36.40Qwell[]You *can* however, plug a bunch of phones into one FXS
19:36.40TripleFFFFgot a weird one.. CDR's  are messed, when sipuser1 calls sipuser2 wich has a forward on his phone (302) it sends to Local/${EXTEN}@context ... but i dont see the original call flow..
19:36.41shellsharkrr--: FXO comes from the Office (as in, central office), FXS goes to a Station (as in phone)
19:36.49Qwell[]but, of course, they'll all ring simultaneously
19:37.05Qwell[]it's like plugging in 5 phones in your house to one Bell line
19:37.18Dr-Linux|homeQwell: so does 1.4beta version sovles my chan_sccp my problem?
19:37.25Qwell[]Dr-Linux|home: no, chan_sccp doesn't work in 1.4
19:37.36Dr-Linux|homeehh
19:37.39Qwell[]Sergio isn't fixing it, and nobody wants to fork chan_sccp
19:37.57Dr-Linux|homeQwell[]: but my phone wants only
19:38.02hbsmurfQwell[]:  chan_skinny is being supported though, right?
19:38.06Qwell[]hbsmurf: correct
19:38.11Qwell[]well...  "supported"
19:38.18Zodiacali guess thats a no for speech-to-text in asterisk..
19:38.19Qwell[]I have NO idea if it's supported in BE
19:38.21De_Monwill app_managerredirect work in 1.2 without any tweaking?
19:38.30Qwell[]Zodiacal: lumenvox
19:38.37shellsharkZodiacal: there is Sphinx, but that only does speech rec AFAIK
19:38.46Dr-Linux|homeQwell[]: huh? who come chan_skinny is supported for my conference phone?
19:38.52Zodiacalyeah speech rec/text
19:38.54Qwell[]Dr-Linux|home: because nobody has sent me one
19:38.57Zodiacalis there a difference?
19:38.59Qwell[]if I had one, I could fix it
19:39.03Zodiacalqwell shellshark Thank You!
19:39.04Qwell[]Zodiacal: huge difference
19:39.15Qwell[]speech recognition == grammar recognition
19:39.19hbsmurfQwell[]:  So, are you the skinny guy?
19:39.27Dr-Linux|homeQwell[]: but normally chan_skinny doesn't work? right
19:39.32Qwell[]speech to text will make every word you say, become text.  That's like Dragons Naturally Speaking
19:39.36hbsmurfQwell[]:  Do the button labels for the 7920 work in 1.4?
19:39.40Qwell[]hbsmurf: I am
19:39.47Qwell[]button labels?  soft keys?  yeah, most of them
19:39.53Qwell[]Dr-Linux|home: doesn't work?
19:39.54Zodiacalqwell speech to text is good then.. have you guys tried them? are they any good?
19:40.08Qwell[]lumenvox does speech recognition, not speech to text
19:40.11Zodiacalwondering if its easy to make a directory voice activated
19:40.16Dr-Linux|homeQwell[]: doesn't wory with my cisco confernece phone?
19:40.17filespeech to text requires training to become accurate, while speech recognition is speaker independent and doesn't require training
19:40.18Qwell[]but, really, why would an IVR need speech to text?
19:40.18Dr-Linux|homei meant
19:40.20filemay require tweaking mind you
19:40.21b11di also would like to do that Zodiacal
19:40.34hbsmurfQwell[]:  Cool.  I've got a 7920 sitting here along with a bunch of 7960s and 7910s.  I've also got a Call Manager Express system in the rack behind me turned off.  :)
19:40.50Qwell[]Dr-Linux|home: no, it doesn't.  not until somebody sends me one, OR sends me a dump from a newer version of call manager for that phone
19:41.05Zodiacalqwell doesn't need it, but it would be nice
19:41.10Zodiacaleasier to find someone
19:41.33b11dthe new design of microsoft.com licks unpleasant things
19:42.00linageei think i got scam called. incoming caller id: 212-440-9180
19:42.27linageethey identified themselves as AT&T yellow pages. i am on the line with them right now and they verified i don't even have an account with them....
19:42.33hbsmurfmicrosoft changes their web page design every few months
19:42.33Dr-Linux|homeQwell: how can i send your dump from a newer version of call manager?
19:42.38hbsmurfI hate it
19:42.45b11dwatch it.. theres another org called "Yellow Book" that does that shit
19:42.46DirtyDCan asterisk detect sound by pitch?
19:42.49PupenoRshould Set(MONITOR_FILENAME=...) to record queues go in queue.conf or somewhere else ?
19:42.52b11da friend of mine got fucked over by them
19:42.53hbsmurfYellow Book is scary
19:42.56Dr-Linux|homeQwell[]: i can have all cisco versions
19:43.02hbsmurfOur local rep yelled at my wife when she cancelled our ad
19:43.04linageeb11d: wtf. they said at&t. that would just be deceptive.
19:43.08*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
19:43.09b11dyep..
19:43.14linageeb11d: i called the number back and i get a busy tone
19:43.19b11dthats suspect.
19:43.25hbsmurfIt's probably an Asterisk call center installation
19:43.25hbsmurf:)
19:43.27b11d:)(
19:43.28b11dbrb
19:43.31linageeb11d: it looks like a NY number and verizon service. (i'm in calif)
19:44.19*** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net)
19:44.26Dr-Linux|homehbsmurf: what firmware you are using on 7920?
19:44.50DirtyDIf I wanted to play call an asterisk extension, and have the asterisk box somehow detect a detect a pitch by frequency, is this possible?
19:45.16b11dhow do you detect pitch via frequency/
19:45.25b11dor maybe I dont know enough about how sound works.
19:45.39hbsmurfDr-Linux:  uh
19:45.46hbsmurfDr-Linux:  lemme check
19:45.51hbsmurfDr-Linux:  firing it up
19:46.02DirtyDb11d, what I mean is can asterisk sense tones, besides DTMF.
19:46.03*** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-a58807a008bab31f)
19:46.03hbsmurfDr-Linux:  my 1.2.13 install will die when it registers
19:46.26hbsmurfDr-Linux:  7920.3.3-01-06
19:46.39Dr-Linux|homehbsmurf: i believe that phone is not sip supported, correct?
19:46.59b11dohhh.. im sure it can, somehow..
19:47.01DirtyDIf I played a sound at 2599 hz,can it determine that the tone is a 2599hz tone?
19:47.02hbsmurfDr-Linux:  Correct.  Skinny only
19:47.07b11dI dont know how to do that.. good question though
19:47.12*** join/#asterisk findlay (n=justin@67.137.24.115)
19:47.18hbsmurfDr-Linux:  I've used 7910s and 7960s with good luck on earlier Asterisk versions
19:47.23b11di wonder if you'd have to record, pass it to an AGI script that checked it with mpg124 or sox or something..
19:47.23Dr-Linux|homehbsmurf: can i /msg you?
19:47.26b11dand then returned the result
19:47.28hbsmurfDr-Linux:  But only with chan_sccp
19:47.32hbsmurfDr-Linux:  Sure
19:47.35Dr-Linux|homethanks
19:47.54*** join/#asterisk dmd1222 (n=opera@c-24-20-35-49.hsd1.mn.comcast.net)
19:48.09DirtyDb11d, how does it go about detecting DTMF? OR is that done by the FXO?
19:48.21b11dI dont know the answer to that, im sorry to say.
19:48.26b11dI dont know where thats handled
19:49.01b11dyou're in the right place to find out though
19:49.17*** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net)
19:50.22Dr-Linux|homei'm using about 40 cisco 7940/60 since one year with no problem. My asterisk 1.2.0 uptime is 246 days
19:50.25b11dsomeone needs to invent a peppermint, that when you eat one, you dont need to continue to eat them all day to avoid that horrible aftertaste.
19:50.35hbsmurfnice
19:50.37b11d246 days of securiy holes.. nice..
19:50.40b11d:)
19:50.42hbsmurfI screw with my asterisk too much to have that kind of uptime
19:50.44DirtyDI guess that's all done inside the channel driver
19:50.49hbsmurfheh
19:50.52b11duptime is a fucking joke.. no one should be proud of uptime..
19:50.58b11dit means you didnt patch.. and you didnt upgrade
19:51.01Dr-Linux|homehbsmurf: ohh your nick is not registerd?
19:51.09hbsmurfI don't think so
19:51.10hbsmurfI'll fix that
19:51.31Dr-Linux|homehbsmurf: i can't see your any pvt message or you never sent me a message in pvt :S
19:51.33hbsmurfthere
19:51.38b11dwoah.. mints and folgers coffee taste like those mint girl scout cookies
19:51.45hmmhesayshehe
19:51.49hbsmurfI've been sending you pms
19:51.53hbsmurfhow about ow?
19:52.06b11dhey hmmhesays..  
19:52.12b11dhow were you going to handle outbound faxing?
19:52.14hmmhesaysif you're not registered your pm's go nowhere
19:52.27hmmhesaysb11d: ip to pstn?
19:52.37b11dwhat is in that process though?
19:52.43b11dyeah
19:52.56hmmhesaysright now i'm using stun on my ata's  for that nat, and redirects directly to my terminating gateways
19:52.57b11ddo people scan documents and then email them to a phone number or something liek that, or what?
19:53.09hmmhesayst.38
19:53.17b11dI need to look into t.38, you're saying?
19:53.52ChkDigitHylafax does the prior.
19:53.59hbsmurfHylafax owns
19:54.28b11dI havent used hylafax in years
19:54.44hbsmurfIt's quite nice
19:54.54hbsmurfThe windows client is a bit clunky,  but it works nicely
19:55.02hbsmurfand it's way cheaper than $2500 for fax server software
19:55.02hbsmurf:)
19:55.56mercesteshylafax ftw
19:56.04b11dfuck the world?
19:56.21mercestes*shrugs*  I dunno...I never did figure out what it meant.
19:56.26b11dFor The Win?
19:56.28mercestesit just seems to be a positive reaction so...ftw.
19:56.44*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
19:57.31mercestesmy ex girlfriend ftw....that's why she's my ex
19:57.39b11dlol
19:58.01hbsmurfex-girlfriends are always ex-girlfriends for a reason
19:58.13b11di have only sex-girlfriends
19:58.55*** join/#asterisk KenSentMe (n=KenSentM@a82-92-80-8.adsl.xs4all.nl)
19:58.57hmmhesaysheh
19:59.10b11dsweet..  the annual faculty xmas party flier says that Paid Escorts will be welcome :)
19:59.30hbsmurfQwell[]:  There are entries in the chan_skinny.c file for the 7935.  Does that mean it will work or just kinda or no way don't try it?
19:59.31b11dI should show up with a gaggle of whores
19:59.38hbsmurfHow many whores is in a gaggle?
19:59.48b11d20+
19:59.49*** join/#asterisk ahigerd (n=ahigerd@adsl-75-19-79-5.dsl.wchtks.sbcglobal.net)
19:59.54hbsmurfThe guy in the cube next to me wants to know too
20:00.00b11dand theres 25 gaggles to a shitload
20:00.16b11da shitload of whores would urle
20:00.19b11durle = rule
20:01.00b11dthat is, unless you're using metric units to meausre whore availability
20:01.04b11dthen we get into decawhores and kilowhores..
20:01.12b11dmy whores are paid by the kilowhore hour..
20:01.18b11dwhat a bad joke.
20:01.19b11d:)
20:01.25hbsmurfROFL
20:01.26linageewhat is the best way to tie a cell phone to asterisk? hrm
20:01.26*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
20:01.53hbsmurfNylon rope>
20:01.53linageei've got it. have asterisk send out a unique caller ID. have the cell phone do a unique ring for that number. have every other number have ring = NULL
20:01.55mercesteslinagee:  Check out Sprint Integrated Office.
20:02.07linageemercestes: what's that? some businessy $$$ thing?
20:02.15mercesteslinagee:  Yea
20:02.31linageeonce asterisk rings the number, you can have festival read out the number and/or name to you when you pick up
20:02.37b11dfestival rules..
20:02.58linageeof course you can do the usual blacklisting. or if no caller ID, ask them for one
20:03.02rudholmlinagee: I do something like that for my front door --the front door speakerphone is on a ringdown line that calls various phones (including my cell) with a unique CID.  The cell associates that CID with a doorbell ringtone.
20:03.16linageerudholm: hehehe
20:03.30Strom_Crudholm: I didn't read your handle and almost responded "hey, my friend has that too!"
20:03.37rudholmhahaha
20:03.53linageerudholm: you can even pick a caller ID that would otherwise never be used (like IP address 10.x.x.x range. :) )
20:03.59linageerudholm: 011-123-1234
20:04.00*** join/#asterisk sloth (n=sloth@64.3.170.41.ptr.us.xo.net)
20:04.05rudholmlinagee: yep
20:04.07*** join/#asterisk TheBearded1_ (n=criggs@203.39.cm.sunflower.com)
20:04.13linageerudholm: oh whoops. just don't hit callback. lol
20:04.22linageerudholm: i'm sure there are other unused spaces
20:04.23rudholmlinagee: in my case, I just send the internal extension to the cell
20:04.36b11danyone ever setup an internal network using the 223.x.x.x address space?  
20:04.37TheBearded1_can anybody tell me how I would know if a sip trunk was registered/connected okay?
20:04.40b11dtheres this place here that did..
20:04.42b11ddoenst make any sense
20:04.46b11dits not RFC1918 compliant..
20:04.56b11dand IIRC, 224.+ is multicast..
20:04.59rudholmno, it's not RFC1918
20:05.14rudholmyep, 224 would be Class D (multicast)
20:05.14b11danyway, I cant wrap my mind around why they did that..
20:05.19rudholmstupidity?
20:05.25b11dmore than likely.
20:05.26linageerudholm: 224 is a nice area code. :)
20:05.34b11dI miss my 807 AC
20:05.40b11d807 4 LIFE!
20:05.41rudholmI miss my 213 AC
20:05.44hbsmurf231 ftw!
20:05.45b11dim 218..
20:05.46b11dso close
20:05.46b11d:)
20:05.49b11dFTW!!
20:05.57Strom_C323 ain't so bad
20:05.57TheBearded1_I'm getting messages like this in my asterisk console: Retransmitting #4 (no NAT) to 65.110.41.100:5060
20:05.59Strom_Cit's symmetrical
20:06.05rudholmyeah, but 323 should have been an overlay!
20:06.06Strom_Cer, palindromic
20:06.10b11d323 is cool.. I used to have a 323 as the local CO code..
20:06.14b11di dont know what thats called
20:06.16linageewhat message do you get when you call a number that's in non existant space?
20:06.18b11din fact, what IS that called?
20:06.22Strom_C2131310
20:06.23b11d<PROTECTED>
20:06.28b11dwtf is each technically called?
20:06.30rudholmmy parents had 213 265 XXXX for 30 years, now that number is a law firm downtwon.
20:06.31*** join/#asterisk DaeJeon-Newbie (i=Singh@124.62.150.38)
20:06.41DaeJeon-Newbiehello guys
20:06.42linageeb11d: areacode-prefix-????
20:06.46b11dyeah..
20:06.47Strom_Cb11d: numbering plan area, office code, line number
20:06.50ahigerdSeveral questions... First off, what's it called when you've got two phone numbers coming in on one phone line?
20:06.51b11dthanks Strom_C
20:06.51linagee???? = "other numbers" :)
20:07.02Qwell[]also NPA-NXX-
20:07.04rudholmon btw Strom_C, I found the 213 -> 310 Press Release I was looking for the other week
20:07.06linageeStrom_C is the CO mad hatter. :)
20:07.19b11dnice :)
20:07.38Strom_Crudholm: cool
20:07.40mercestesb11d:  Local exchange.
20:07.43ahigerdOne physical phone line, that is; it's two lines and a multi-line phone or fax machine can use it as such.
20:07.50mercestesb11d:  NPA, NXX, PHN.
20:08.04Strom_Cahigerd: "distinctive ring"
20:08.07Strom_Cor "DID"
20:08.11b11dyeah I wanted to call it Local Exchange Code but that'd be LEC and LEC is Local Echange Carrier (i think?)
20:08.14Strom_Cdepending on how it's provisioned
20:08.16b11dExchange that is
20:08.21Strom_Cb11d: prefix or office code
20:08.29b11dcool..  that fits nicely... thanks
20:08.34TheBearded1_can anybody help me debug my sip trunk problem?
20:08.37linageeStrom_C: is it possible some apocolyptic number exists? call it, it goes to another CO, goes to another CO, again and again until it ties up all the circuits? heheheh
20:08.43mercestesb11d:  LEC is Local Exchange Carrier.
20:08.48Strom_Clinagee: no.
20:08.52b11dthanks for verifying that
20:08.52linageeStrom_C: aw. :(
20:09.06b11dTheBearded1_.. do you REALLY have a beard?
20:09.07b11dbecause I do..
20:09.08rudholmyeah, I usually say "prefix" although I know that's kind of ambiguous since a "prefix" is also possibly a vertical service code (or anything dialed at the beginning of a dialing pattern, really)
20:09.11b11dI take the beards seriously
20:09.18mercestesb11d:  It's called NXX because that's the pattern match for it in old switches.  :)
20:09.25TheBearded1_yes i do
20:09.26rwxr--r--b11d www.nanpa.com (helps)
20:09.28b11dcool
20:09.28Qwell[]s/old//
20:09.38b11dthats handy
20:09.46Strom_Cmercestes: and, just for trivia, what was the standard pattern in most area codes before 1995?
20:09.50*** join/#asterisk dmd1222 (n=opera@c-24-20-35-49.hsd1.or.comcast.net)
20:09.54mercestesQwell, True.
20:09.54TheBearded1_b11d: beards are handy?
20:10.06b11dyep
20:10.08b11dso was that link
20:10.11b11dbut yeah.. beards are handy
20:10.20b11din fact, if I can give everyone only one bit of advice.. it's GROW A BEARD.
20:10.21linageebeards?
20:10.22ahigerdStrom_C: Oh, that's how it works? So if I wanted my Asterisk server to only pick up on line 2, I'd have to set it up to recognize distinctive ring?
20:10.27b11d(yes, stolen from Family Guy)
20:10.28mercestesStrom_C:  hm....I'm not sure....
20:10.38Strom_Cahigerd: do you have DID, or do you have distinctive ring?
20:10.41Strom_Cmercestes: NNX
20:11.02rob0Beard is a great place to store food to eat later.
20:11.12ahigerdStrom_C: No clue; I'm actually troubleshooting for a remote customer. ^^()
20:11.16hmmhesaysfamily guy rocs
20:11.16robl^b11d: you must have given advice to my great aunt Ethel.  she's been growing a mustache and beard for years
20:11.18b11dI find the beard DOES limit your choice of women though..
20:11.19b11dsome hate them
20:11.23b11dHAHAHA
20:11.24mercestesyou know...I *have* seen that in some old CSX programming.
20:11.31linageeStrom_C: is it possible to buy the "default extension" for a prefix? AREACODE-PREFIX-nonexistant
20:11.32Strom_Cahigerd: well, then find out
20:11.33ahigerdBut that gives me a question I can ask; thanks.
20:11.37hmmhesaysmy friend was hitting on a chick with a beard wednesday night
20:11.43Strom_Clinagee: what do you mean
20:11.43mercestesnow it makes sense..:)
20:11.43TheBearded1_b11d: if they don't like beards you don't want them anyway, look at it as a bult-in filter for the bad ones
20:11.59b11dI do.. believe me ;)
20:12.07linageeStrom_C: when the message that you've called a non existant number comes on. is it possible for instead of that message for the call to get routed to you? heh
20:12.08DaeJeon-NewbieI am unable to install libstdc++6 -packeage using yum-centos. is there any other way?
20:12.17*** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net)
20:12.29b11dwww.cnac.ca
20:12.30danpbeautiful: http://flickr.com/photos/dpiddy/322743760/
20:12.31b11dnow here we go
20:12.33Strom_Clinagee: you'd have to register as a CLEC, buy a switch, and have your own office code assigned by NANPA
20:12.39linageeStrom_C: or wait. n/m. if you owned the entire prefix, you could decide what to do
20:12.41linageeStrom_C: jinx. :)
20:12.57linageeStrom_C: replace the word "prefix" with the proper terminology
20:13.12*** join/#asterisk dasenjo (n=dasenjo@190.24.176.58)
20:13.20Dr-Linux|homegoogle likes my site for asterisk stuff :P
20:13.27Strom_Clinagee: it sounds to me like you've read just a few too many phreak kiddie text files
20:13.37robl^what's the going rate for a block of 100 DIDs for US?
20:13.41linageeStrom_C: way long time ago maybe
20:13.44TheBearded1_b11d: know asterisk sip trunks well?
20:13.54linageerobl^: a bj
20:13.56linageelol
20:14.07hmmhesayssome chicks like the way the beard tickles their thighs
20:14.07TheBearded1_robl^: at&t gives blocks of 20 at $3.99
20:14.08*** join/#asterisk sloth_ (n=sloth@64.3.170.41.ptr.us.xo.net)
20:14.11b11dI've done some work there
20:14.22robl^linagee: oh?!?!?!  hrmmm
20:14.31linageeTheBearded1_: do you have to pay once or every month or what?
20:14.40TheBearded1_linagee: yep, monthly
20:14.56linageeTheBearded1_: i think there's some DID rule about if you don't use it, you have to return it to the DID population. (CLEC rules)
20:15.09hmmhesaysor pay more to have it idle I think
20:15.17robl^now if I can find a SIP or IAX provider that would give me a block of 20 (or 100)  DIDs
20:15.22b11dahh nothing like box
20:15.25Strom_Cfairly sure that rule only applies to prefixes
20:15.33linageeStrom_C: true
20:15.41hmmhesaysi love this dell commercial
20:15.51rwxr--r--i wonder w/ i hate more cisco or ms
20:15.51linageehmmhesays: "Dude, get a dell"?
20:16.14b11danyone see the movie "The Jerk" ?
20:16.14hmmhesaysguys calls in to dell "i'd like to get computer for my daughter" - "ok sir how do you think she will be using it" - " well I think she said something about a showercam site"
20:16.26linageeStrom_C: btw, what is "get a phone switch" these days? hehehe. sure you can buy a $100,000 piece of equipment, but can asterisk talk "get a phone switch" yet?
20:16.28b11dlol
20:16.29hbsmurfI'd like a subscription
20:16.30hbsmurfplease
20:16.38linageeStrom_C: with the appropriate cards
20:16.50linageeStrom_C: SS7, right?
20:16.50Strom_Clinagee: I don't believe you can do all the necessary class 5 stuff with asterisk yet
20:16.53linageehm
20:16.57b11dthat reminds me.. I need to go take care of that GD CPE light on my T1
20:16.59Strom_CSS7 is one part of it, yes
20:17.00hmmhesaysi'd whore out my gf to a showercam site if I got part of the cash
20:17.07linageeStrom_C: interesting
20:17.16b11dlets talk about that a little bit hmmhesays..
20:17.18b11d:)
20:17.25hmmhesayslol
20:17.47linageeStrom_C: you know when you gave me all those links to those CO resources, i was working 80 hour weeks. i have more free time now though. :)
20:18.32linageehmmhesays: gf?
20:18.49hmmhesaysgirl friend
20:18.56linageehmmhesays: gf = guyfriend
20:19.04hbsmurfExcuse me, let's get back to the showercam subject please
20:19.09hmmhesaysLOL
20:19.12TheBearded1_hehe
20:19.13TheBearded1_Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 65.110.41.100
20:19.26TheBearded1_wild guess here, but, not a good thing?
20:19.58robl^linagee likes the rough stuff
20:20.05hmmhesaysoh i love fridays
20:20.12rwxr--r--thebearded... I got a ranDumb sip message generator
20:20.22linageerobl^: iron cooking pan beats a cup
20:20.25rwxr--r--the normal SIP error 500 should be internal server error
20:20.28mercestesI'd whore out my g/f for props.....*sighs*  I'm a bad person.
20:20.31rwxr--r--what gave you that message?
20:20.32hmmhesaysi'm going to do nothing today except play guitar
20:20.48linageehmmhesays: is that what they're calling it these days. :)
20:20.50TheBearded1_i got that from fonosip.com
20:20.58rwxr--r--k thanks
20:21.09hmmhesaysi'm also going to play star wars battlefront 2
20:21.12hbsmurfIs this #asterisk or #whoreoutyourgirlfriendforvoip?
20:21.16TheBearded1_rwxr--r--: preceded by: SIP/2.0 500 could not create new transaction
20:21.34hmmhesaysi'm in the middle of taking hoth
20:21.36rwxr--r--strange not standard error code
20:21.38rwxr--r--oh well
20:21.41TheBearded1_hbsmurf: vonage accepts females for payment now?
20:21.41*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)
20:21.45robl^anyone know how to go about getting a contiguous block of DIDsfrom eith an IAX or SIP provider?
20:21.46rwxr--r--www.infiltrated.net/slicer
20:22.02linageeTheBearded1_: if vonage accepted paypal and paypal accepted foreign currency, then yes.
20:22.17robl^what about those who whore out their boyfrieds or small farm animals?
20:22.23*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
20:22.26hbsmurfTheBearded1_:  I think you'd need more than 1
20:23.15*** join/#asterisk sloth_ (n=sloth@64.3.170.41.ptr.us.xo.net)
20:23.32TheBearded1_rwxr--r--: would you guess it's fonosip's fault or mine?
20:23.40linageewake me up when "expensive phone switch" = asterisk running inside of a 4 cpu box with special cards
20:23.43mercesteslinagee:  What is the exchange rate on the US dollar to foreign females?
20:24.04b11dok..  big gay asterisk orgy.. this weekend..  Minneapolis.
20:24.10b11dno names, no questions
20:24.11filetry to stay on top
20:24.15fileat least during the day
20:24.18fileon topic
20:24.18hmmhesaysLOL
20:24.19fileON TOPIC
20:24.26hmmhesaysnice file nice
20:24.28robl^on top!?!?!?
20:24.30linageefile: "try to stay on top" = you're going? lol
20:24.31fileI hate you all :P
20:24.34b11dhahaha
20:24.35linageelol
20:24.38mercesteslol
20:25.03b11dI thought it was on topic.. I said asterisk..
20:25.06b11dmy apologies
20:25.06Kattyi'mmmmmmmmmmmmmmmm goign insane.
20:25.07rwxr--r--file www.infiltrated.net/hate.jpg makes a great desktop
20:25.13Kattyfile: i had a nap on my desk!
20:25.16mercestesI love you, Katty
20:25.17linageeb11d: you also said Minneapolis
20:25.18fileKatty: was it comfy?
20:25.23b11dthat I did
20:25.44Kattyfile: not so much, no
20:27.32*** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com)
20:27.47b11dwell im going to go try to take care of this CPE alarm..
20:27.47b11dbbl
20:28.16b11dok, you can help.
20:28.34mercestesyay!  :D  thanks.
20:28.38b11d:)
20:28.40mercestespaint it green.
20:28.45b11ddone and done
20:28.53b11dbtw, ever watch Star Trek, TNG much?
20:29.11b11dever notice that when the ship is really fucked.. Geordi LaForge opens that little panel to the side of the warp core?
20:29.14linageeStrom_C: say you have a big enterprise telephone switch that talks all the right languages. how do you get the telephone network to start sending you prefixes? is it hella expensive?
20:29.19b11dI want one of those panels in my data center
20:29.24linageeStrom_C: does it involve golf games?
20:29.29b11dso I can crack it open and yell "We've got a coolant leak!"
20:30.07fileKatty: I talked to this person on the phone a sec ago... she said her name was Katty - any relation?
20:30.10mercestesb11d:  that would be awesome.
20:30.14b11dyeah, I think so :)
20:30.20*** part/#asterisk TheBearded1_ (n=criggs@203.39.cm.sunflower.com)
20:30.27mercesteskatty:  can I call you on the phone?
20:30.27Kattyfile: ummummumm.
20:30.28*** join/#asterisk Paavum (n=chiardon@200.71.58.39)
20:30.33Kattymercestes: mayhaps
20:30.34b11dand you guys think im fucked
20:30.35mercestesplease?
20:30.38hbsmurfYOu aren't?
20:30.43b11dI am..
20:30.44hbsmurfI love easy jobs
20:30.48hbsmurfCLICK
20:30.50hbsmurfThat's $50
20:30.51hbsmurfthanks
20:31.06hbsmurfSo
20:31.09b11di'll split that with you
20:31.10hbsmurflet me throw this at you
20:31.13PaavumHi
20:31.15Kattymercestes: are you canadian?!
20:31.16hbsmurf300 user school system
20:31.18rwxr--r--$50 is not enough for gas sometimes
20:31.25Kattymercestes: 'merican?
20:31.32mercestesKatty:  Which one gets me your phone #?
20:31.32hbsmurfser or asterisk/realtime/dundi?
20:31.35PaavumCananybody tellme if asterisk supports "Shared IP extensions"?
20:31.35rwxr--r--contractor... $200 ph 2 hr min
20:31.38b11dthats about what im rolling out here right now..
20:31.44b11da little over 300 end points..
20:31.45Kattymercestes: i don't give out phone numberseseseses
20:31.49b11dim running straight asterisk on FreeBSD
20:31.51mercestesaw..:(  I'm 'merican.
20:31.56Kattymercestes: but i do have a lil iax number in my pocket.
20:32.02mercesteswoohoo!
20:32.07b11dI'm dual citizen :)
20:32.14hbsmurfStraight asterisk?
20:32.19mercestesI'm a canadian spy living under cover in America.
20:32.21b11dyeah..  not abe, or realtime, or whatever..
20:32.21hbsmurfMultiple servers
20:32.22hbsmurf?
20:32.23Kattyhmmhesays: and you, sir, have been a snob.
20:32.25hbsmurfah
20:32.28Kattyhmmhesays: not a single call!
20:32.31b11d3.. syncing every minute..  
20:32.34hbsmurfload balancing or anything?
20:32.36b11dless than a second of downtime when I pull the plug
20:32.37hbsmurfhow are you syncing?
20:32.43b11drsync & pfsync
20:32.49b11dand CARP for failover
20:32.54hbsmurfdo the endpoints register to different boxes?
20:32.59PaavumI want to  use asterisk with the Polycom 601
20:33.03b11dsince the astdb is synced all the time.. its not necessary
20:33.14hbsmurfhow do you sync astdb?
20:33.20b11dthe call will be cut, but you can redial instantaneously
20:33.24b11dwith rysnc..
20:33.26hbsmurfright
20:33.27hbsmurfah
20:33.28hbsmurfok
20:33.38b11dim really happy with it..  so far..
20:33.46b11dbut keep in mind..
20:33.47*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.197.Dial1.SanJose1.Level3.net)
20:33.56hbsmurfSo, are you registering to one server or multiples?
20:33.58b11dwhile I have over 300 endpoints, not more than say.. 19 have ever been in use at the same time
20:34.10b11djust to the single server
20:34.14hbsmurfok
20:34.15hbsmurfI get it
20:34.18hbsmurfWith standard asterisk
20:34.19hbsmurf1.2.13?
20:34.20b11dyep
20:34.25b11di'll go to 1.4 when the time comes
20:34.26hbsmurfwhat kind of hardware?
20:34.40b11ddual opteron 1.8Ghz boxes on Tyan Mobo's
20:34.44b11d2gb of RAM each..
20:34.44hbsmurfreally?
20:34.45hbsmurfWow
20:34.45b11dyeah
20:34.49hbsmurfThat's awesome!
20:34.58b11dis it?  why>?
20:35.04Strom_Cyay, my 480i is here
20:35.06b11dI've got a 400GB RAID on each for VM.
20:35.14hbsmurf400GB?  
20:35.15hbsmurfCrap
20:35.19*** join/#asterisk matt__ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d)
20:35.19b11dyeah its probably overkill
20:35.22hbsmurfThat's a huge amount of voicemail
20:35.32b11dyeah..  but I want the system to be there in case its necessary
20:35.34hbsmurfI was going to go with like 60gb for the school I'm quoting
20:35.37*** join/#asterisk matt_ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d)
20:35.45b11dit really boils down to:  how fast can you get replacements?
20:35.58b11di can get 200's quicker than 60's.. but thats just how things are here.
20:36.10hbsmurfI sell HP hardware, I can get stuff in less than a day
20:36.11hbsmurf:)
20:36.21b11dahh.. these are all homebrewed
20:36.22mercestesHP blows goats.
20:36.29hbsmurfHP owns!
20:36.31mercestesKatty!  I wanna call joo..:(
20:36.32b11downs nothing..
20:36.33b11d:)
20:36.35hbsmurf:)
20:36.36mercestesHp is teh sux.
20:36.40b11dFTW!
20:36.40hbsmurfNah, I love HP
20:36.45hbsmurfTrained and certified on their hardware
20:36.48danpi'm a big fan of HP procurve switches
20:36.54Kattyoh
20:36.56Kattyk
20:36.57hbsmurfYou spend a week in Wisconsin in the middle of winter and tell me it isn't fun!
20:37.02hbsmurfHP switches are teh such
20:37.03hbsmurfsuck
20:37.04mercestesmy ex wife works for HP...believe me...I know when something blows and somethign doesn't.
20:37.09b11dlol
20:37.11mercestesand my ex definately does not..and hp definately does.
20:37.31b11danyway, I wish I could set up my system to not even drop the calls when the servers roll over..
20:37.33b11dbut I cant yet
20:37.34rwxr--r--hp blows
20:37.40Strom_Calmost as much as your spelling of "definitely"?
20:37.41Strom_C:)
20:38.20b11dhewlett-paqard..
20:38.25b11dsince they bought compaq and all
20:38.45mercestes...  I hate you Strom!  (hugs)
20:39.06b11dwhat kind of phones are you gonna roll out hbsmurf?
20:39.07mercesteshp makes a good printer tho.
20:39.16b11dI went with all poly 501's in the faculty & staff offices
20:39.19hbsmurfIf you know how to set the servers up they're quite nice
20:39.20b11d301's in unused rooms..
20:39.22hbsmurfDamn near bulletproof
20:39.24b11d601's for secretaries
20:39.38b11dyeah I like hp printers but I hate jetdirect..
20:39.42hbsmurfTeh 601s are sweet
20:39.43b11dor.. webjetadmin anyway
20:39.48KattyStrom_C: come back to me!!!
20:39.50*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
20:39.51hbsmurfbut a 601 with three sidecars is almost $1k
20:39.56b11dwow..
20:40.02b11dsingle sidecar here for one user..
20:40.05hbsmurfMy bald headed salesman thinks you're an idiot if you hate jetdirect
20:40.05hbsmurf:)
20:40.18b11dI dont hate jetdirect.. just the admin tools HP provides..
20:40.20Kattysomeone should host an iaxy comference.
20:40.25Kattyso we can all chitchat.
20:40.26Kattythat'd just be peachy
20:40.27hbsmurfOh yeah, the admin tools suck
20:40.31b11dI hate the god damn jetdirect interfaces that you cant program right on the printer
20:40.34Kattyfile: set one up!
20:40.38hbsmurfI need to be able to send messages to the dman Polycom phones
20:40.39danpi am really liking my 601's...i have 3 sidecars for 3 phones but two of those will probably be replaced with te flash operator panel
20:40.44hbsmurfIf I could do that I'd be happy
20:40.52b11dyeah I wish I could do that as well
20:40.52Strom_CKatty: I could probably do one here
20:40.59KattyStrom_C: yum.
20:41.05b11di'd like to figure out how to make my asterisk setup act to reinforce the campus emergency alert system.
20:41.08Strom_Cgive me one second
20:41.13hbsmurfExactly
20:41.15KattyStrom_C: i'll come talk
20:41.15hbsmurfThat's what I need
20:41.25hbsmurfthere has to be a way to get messages to pop up on those things
20:41.28b11dI've got it to ring every phone on campus & play a message..
20:41.30b11dbut its not working just right yet
20:41.41*** mode/#asterisk [+o mog] by ChanServ
20:41.41hbsmurfMight be time to call Polycom and use that special support I'm afforded for taking two 5 minute tests
20:41.58fileKatty: OOH OKAY
20:42.00b11dyou should be awarded it for buying their products
20:42.16hbsmurfEh, the tests took a few minutes
20:42.20hbsmurfI earned that support!
20:42.21hbsmurf:)
20:42.22b11d:)
20:42.23wunderkinhbsmurf, you are polycom certified?
20:42.26hbsmurfYes
20:42.27b11dwhen you figure it out. let me know..
20:42.30hbsmurfI will
20:42.32b11dthanks
20:42.35Kattyi wanna be polycom certifimicated.
20:42.38hbsmurfI'm going to call and bitch at them
20:42.59b11dI would too
20:42.59mercestesI passed the polycom certification tests.
20:43.01hbsmurfPolycom certimification is easy
20:43.01hbsmurfeasy easy easy
20:43.01Kattypro-bably.
20:43.01hbsmurfregister for an account, study, take open book test
20:43.02mercestesyea, my pe tmonkey took it for me.
20:43.02hbsmurfDONE!
20:43.02b11dtell them Kimberly Clark pulp products are superiour to the Proctor & Gamble ones they use now..
20:43.06hbsmurfrofl
20:43.10hbsmurfI'll do that
20:43.13b11dplease do :)
20:43.16fileKatty: are you READY for the conference info?
20:43.22wunderkini should probably do that, ive been having some problems with some ip430s... intermittant rebooting.. been waiting forever on a reply from polycom
20:43.27hbsmurfBUt first, I need to bitch at Websense
20:43.31Kattyaree you readdddyy too iaxxxxxxxxx
20:43.36Kattyfile: also! yes.
20:43.36hbsmurfwunderkin: What firmware?
20:43.41filewell I'm not, so gimme a min or 2
20:43.49Kattykay
20:43.58Kattyactually, i have one.
20:44.00hbsmurfPolycom really screwed the pooch with their first released of the sip 2.0 firmware
20:44.03Kattyit's already setup and stuff
20:44.04hbsmurf2.03 is much nicer
20:44.07b11d2.0.3 seems to be good
20:44.11Kattybut i'm lazy
20:44.14filemine is setup too
20:44.18hbsmurfPresence monitoring didn't work and they told me it was Asterisk, not them
20:44.20Kattyhot.
20:44.25hbsmurfReally?  Then why the fix so soon?
20:44.32b11dI dont understand how im going to manage ALL the config files when it comes time to upgrade SIP again on these phones
20:44.44fileIAX2/guest@neutrino.file-radio.com/399 or SIP/399@neutrino.file-radio.com
20:44.44fileGO!
20:44.50hbsmurfI've got most of the config in the sip.cfg and just a little bit in the phone config
20:44.51Kattywooooooo
20:44.58hbsmurfQUICK, DIAL IN!
20:45.01Strom_Cfile: dammit, you beat me
20:45.02wunderkinhbsmurf, i ran this through polycom god ([TK]D-Fender) and he said to rma... but.. im still testing the others... ive had it with 1.6.7, 2.0.1, and 2.0.2... sometimes the phones reboot even when not being used... when mine does it (at home) it reboots after every call, sometimes it is ok for a few days and then starts again
20:45.11b11dI've got like 3 or 4 files per phone on the provisioning server..
20:45.15b11dand with 300 phones..
20:45.17hbsmurfWow
20:45.17b11dyou can imagine..
20:45.22hbsmurfThe 430s are crap?
20:45.22b11dits sick. and difficult to manage..
20:45.24hbsmurfI hope not
20:45.34hbsmurfThey're cheap and perfect for the school I'm quoting now
20:45.34fileStrom_C: you're joining in too? uh oh
20:46.12hbsmurfCELLO AND VIOLIN HEAVY METAL
20:46.14hbsmurfYEAH
20:46.15b11dyou all need to spend time reminding me to eat food..
20:46.17hbsmurfSorry
20:46.20hbsmurfEat
20:46.24b11di dont eat breakfast or lunch anymore ;(
20:46.25b11dNot good.
20:46.31hbsmurfWe just got a box full of cookies from the leasing company we use for our clients
20:46.32hbsmurfI'm fat
20:46.32Katty:<
20:46.38Kattymore people need to be in the conference.
20:46.42b11di'd join but you hate me..
20:46.43hbsmurfMore?
20:46.43b11dso..
20:46.48b11dplus theres that restraining order
20:46.51mercestesI'm at work right now..:(  *cries*  can I do it after work?
20:46.58hbsmurfI spent an hour on the phone with digium through iaxtel today, my voip minutes are burned up
20:47.26wunderkinhbsmurf, you can call digium directly through misery... much better.. check the default extensions file
20:47.35b11dMisery Incorporated..
20:47.41b11ddammit
20:47.42b11dno.
20:47.43b11dno
20:47.47b11dnot this song..
20:47.48hbsmurfI'm being a smart-ass, iaxtel worked fine for me
20:47.48hbsmurf:)
20:47.55hbsmurfI'm just tired of being on the phone
20:48.03wunderkinyeah, but iaxtel can suck a lot.. im sure
20:48.14hbsmurfActually, it sounded great
20:48.27hbsmurfWe've got 4.5Mbps both ways here so I'm sure that helps
20:48.40b11dim going to DDOS you
20:48.46rudholmI was at LISA last week and one of the authors of SIP gave a talk.  He gave his SIP address in the talk, but it didn't work (I was hoping it'd hit the cell on his belt :) )
20:48.47b11dphear my multiple OC3's
20:49.05rudholmonly 4.5MBPS?  pshaw!
20:49.22hbsmurfWell, I don't pay for it so 4.5mbps is fine with me
20:49.26hbsmurfMy landlord pays for it
20:49.30b11dit's plenty fast..
20:49.35rudholmI don't pay for my bw, either, my employer does :)
20:49.43hbsmurfBefore they moved their damn web real estate tour company to Rackspace we were going to go DS3 at 10Mbps
20:49.44b11deveryone needs to remember what it was like five years ago
20:49.51hbsmurfstupid idiots downstairs killed that
20:49.56hbsmurfI remember what it was like
20:49.56b11dtaxpayers of MN pick up mine
20:50.02hbsmurfI ran a dial up isp at my last company
20:50.04hbsmurfISDN was fast
20:50.08b11d:)
20:50.12rudholmI miss my BRI
20:50.13rudholm:(
20:50.13hbsmurfISDN WAS BLAZING FAST!
20:50.16hbsmurfYeah, me too
20:50.20hbsmurfI had a Cisco 804
20:50.28hbsmurfIt would drop a channel when someone called my house
20:50.43rudholmI *really* want to pull a BRI into my home Asterisk box (I just can't seem to get rid of echo issues on my POTS line)
20:50.54hbsmurfWe had a BRI up until last March
20:50.59b11dI have a minor echo entirely within VOIP
20:51.00hbsmurfSBC said they won't sell them anymore
20:51.02b11dit freaks me out a bit
20:51.03rudholmbut the Digium card only supports Euro ISDN, afaik
20:51.07hbsmurfI was running the BRI into a Cisco Call Manager Express system
20:51.28hbsmurfI still hve that system in the rack behidn me
20:51.37rudholmoh, I'm in Los Angeles, AT&T/SBC still sells them here, even in "Residential" form.
20:51.38hbsmurfWe yanked it out and switched to Asterisk
20:52.08Qwell[]if you yanked it out, how is it still in the rack behind you?
20:52.23hbsmurfBecause my desk is in the tech room
20:52.29hbsmurfand my servers are in the server room downstairs
20:52.30hbsmurf:)
20:52.54hbsmurfMy server room has air and generator power
20:52.57hbsmurfand we don't do crap in this building
20:53.17*** join/#asterisk andresmujica (n=AndresMu@190.24.71.182)
20:53.40b11dwhat's that new replacement for HALON?
20:53.42hbsmurfI have a 2610xm, a 3640, a 1602 and a 2524 in the rack behind me
20:53.47b11dI know that halon is banned now..
20:53.49hbsmurfThey're basically shelves now
20:53.57rudholmI think BRI is a great solution for low-density digital entrance facilities
20:54.04hbsmurfI agree on the BRI stuff
20:54.09hbsmurfIf only they could do name caller id with it
20:54.11rudholmfor small businesses and stuff
20:54.13rudholmyeah
20:54.14monstedwell, that old crap isn't worth anything but being shelves anyway
20:54.18rudholmamen to that!
20:54.21hbsmurfheh
20:54.27rudholmyou can only get CNAM on PRI, dammit!
20:54.29hbsmurfI actually like my analog lines
20:54.30rudholmso annoying
20:54.30b11dmy new APC UPS sits on my old AS/400's
20:54.31hbsmurfThey work fine
20:54.34hbsmurfdigital is so much nicer though
20:54.39rudholmreally?  I can't get rid of echo on mine.
20:54.40hbsmurfI wish they would do frac T1s
20:54.46hbsmurfEcho is a level problem
20:54.50hbsmurfI had my levels too high
20:54.52rudholmyeah, frac T1 ISDN would be great
20:54.55hbsmurflowered them and now no ehco
20:54.58hbsmurfuse ztmonitor
20:55.01rudholmyeah, I went through the whole deal with my levels
20:55.04rudholmyep
20:55.04hbsmurftry to get things around the halfway point
20:55.05rudholmdid that
20:55.07b11d"listen to the sustain.. can you hear it?  wahhhahahhhh.. "
20:55.10hbsmurfI actually used a fax machine to set the levels
20:55.14b11d"yeah but its the sustain.."
20:55.14rudholminteresting
20:55.16hbsmurfI need to fix my guitar
20:55.18rudholmhaha
20:55.20rudholmthis one goes to 11
20:55.23b11dYES!
20:55.28hmmhesayswell i just finished battlefront II
20:55.30hbsmurfI'll get my bass out when I get home
20:55.34b11dwhen we need to push it over the cliff, we can go up to 11
20:55.37b11dothers can only go to 10
20:56.00b11dI wont lie.. for a long time.. I thought Spinal Tap was real.
20:56.11hbsmurfSo, why can't I park with Asterisk from the Polycom phones?
20:56.11rudholmpart of my problem might be my loop length (I'm 26,000 feet from the CO)
20:56.18fileStrom_C: .
20:56.23Strom_Cfile: ..
20:56.32fileStrom_C: ....
20:56.37b11dyou guys and your secret code
20:56.44hbsmurf\\\\slashy!
20:56.49Strom_Cfile: ..:..::::.....::::...
20:56.50hbsmurfSecret codes are gay
20:56.54fileStrom_C: *
20:56.58Strom_Coh noes
20:57.00hbsmurfIt's all about the secret secret codes
20:57.11rudholmAsterisk is gay
20:57.11b11dim a Mason.. i enjoy secret codes. :P
20:57.16hbsmurfNow answer this, why can't I park with the park button on my Polycoms?
20:57.18*** part/#asterisk dmd1222 (n=opera@c-24-20-35-49.hsd1.or.comcast.net)
20:57.25hbsmurfWho do I have to send pizza and beer for that to get fixed?
20:57.27b11dpark button?  601s?
20:57.32hbsmurfYes
20:57.38hbsmurfYou can enable the park button
20:57.49Qwell[]hbsmurf: you could send me pizza and beer...doesn't mean it'll get fixed though
20:57.50hbsmurfI want that sucker to park from there and display the parking slot on the phone
20:57.52b11dwhats the difference between park & hold?  parking puts a call into a queue?
20:57.52Kattyhmmhesays: mew?!
20:57.53hbsmurfheh
20:58.02filehmmhesays: you should call in so we can yell at you
20:58.06hbsmurfQwell[]:  I'll send you pizza and beer to get my 7920 working
20:58.07hbsmurf:)
20:58.13Qwell[]it should work with 1.4 :p
20:58.18hbsmurfPark puts the person into a slot
20:58.22hbsmurfso someone else cna pick them up
20:58.27b11dthats what I thought. cool.
20:58.46hbsmurfTransfer 700 - the phone system reads back the slot number - transfer again.  Dial parking slot number somewhre else
20:59.12hbsmurfI love the parking concept, but the implementation is teh suck righ tnow
20:59.17b11dftw!
20:59.26hbsmurfThat and I want a button for recording calls
20:59.33hbsmurfHit the button and Asterisk records
20:59.39hbsmurfThat would be so nice
20:59.45b11dheh.. I want a way to overlay a heavy breathing sound into any call I choose..
20:59.55hbsmurfI think digium is calling me back
20:59.59b11dexcellent
21:00.01b11danswer it :)
21:00.01lters_GotoIf
21:00.02b11dlearn
21:00.08hbsmurfIt's probably sales
21:00.13hbsmurfI called and bitched about the 40 call limit
21:00.14hbsmurf:)
21:00.17lters_I have case where it is returning true
21:00.24lters_and should be false...
21:00.30hbsmurf120 calls in ABE A.2-5, 40 calls in ABE B.1-2
21:00.34*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
21:00.46lters_GotoIf([${VMBOXEXISTSSTATUS} = "SUCCESS"]
21:01.00lters_but evals to true when it the var is false.
21:01.10lters_[FAILED == "SUCCESS"]
21:01.20Qwell[]$[
21:01.38hbsmurfIt IS Digium!
21:01.46lters_does it need the $
21:01.57pollerHi. I'm having a problem. Setup is 1 asterisk with a sip-trunk and a Nokia e60. The outside party can hear me on my nokia but i can't hear them. Any input? Configs: http://tmp.poller.se/sip.conf & http://tmp.poller.se/extensions.conf
21:02.31lters_Qwell[]: thanks
21:02.35b11dthers your problem
21:02.47lters_but one = or 2 ?
21:03.00lters_= or == for an eval ?
21:03.37pollerb11d: Mine? And where? :)
21:04.24hbsmurfWell, that did me no good
21:04.29hbsmurfno academic pricing for abe
21:04.37hbsmurfno clue as to what license levels are offered
21:04.48hbsmurfI'll just go open source
21:05.05b11dsorry poller..  wrong window
21:05.31DaeJeon-NewbieI am trying install a VXML broswer
21:05.49DaeJeon-NewbieRegistered application 'Vxml'  == Parsing '/etc/asterisk/vxml.conf': Found    -- Bad Video codec.
21:05.54pollerb11d: No, now you have to help me. ;)
21:06.00b11dhahaha
21:06.03b11di'll take a  peekl
21:06.17b11done way audio eh..  sure its not a RTP routing issue?
21:06.56hbsmurfLet's talk one way audio for a sec
21:07.02hbsmurfHere's something for you to have fun with
21:07.11hbsmurfset the call waiting beep in the polycom sip.cfg to a 0 length
21:07.46b11dwhat happens?
21:07.46*** join/#asterisk blitzrage (n=blitzrag@dsl017-122-217.mci1.dsl.speakeasy.net)
21:07.46hbsmurfwhen someone calls an extension already in a call the audio stops on the existing call
21:07.47hbsmurfon the inbound leg
21:07.47pollerb11d: The asterisk is behind nat but i forwarded udp/5060,10000-20000.
21:07.47hbsmurfit's fun!
21:07.54hbsmurfI found that out the hard way
21:07.55b11dI dont touch NAT issues..
21:07.57b11d;)
21:07.59hbsmurfAlmost drove me to drink
21:08.08pollerb11d: Hehe, good for you. ;)
21:08.30b11dwell..
21:08.34*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
21:08.39b11dverify the ports that are in use (or attempting to be used) actually fall within that range..
21:08.50b11dget tcpdump or snort or ethereal or something
21:08.53b11dhell.. use rtp debug
21:08.54b11d:)
21:08.59b11dand sip debug of course
21:09.02hbsmurfpoller, you don't have the e60 marked as using nat in your sip.cfg
21:09.05hbsmurfer, sip.conf
21:09.22pollerThe e60 isn't using nat.
21:09.26hbsmurfAh
21:09.28pollerIt's in the same subnet as the asterisk.
21:09.29hbsmurfThat would explain it
21:09.30poller:)
21:09.34*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)
21:09.35hbsmurfbut the sip trunk out isn't working?
21:09.51pollerI don't know what it is that aint working.
21:09.52b11dbest post i've seen all day:  http://pastebin.ca/280350
21:10.16DaeJeon-NewbieQwell:Registered application 'Vxml'  == Parsing '/etc/asterisk/vxml.conf': Found    -- Bad Video codec.
21:10.23*** join/#asterisk CleanerX (n=nix@p54A393D3.dip0.t-ipconnect.de)
21:10.28hbsmurfI'd say your sip trunk isn't working
21:10.31hbsmurfIf the e60 is internal
21:10.33hbsmurfit should work
21:10.36*** join/#asterisk cthorner (n=cthorner@209-234-185-148.static.twtelecom.net)
21:10.44Qwell[]uhh...  vxml?  We don't support that
21:10.45hbsmurflemme chekc my firewall
21:10.50pollervoip*CLI> sip debug
21:10.50pollerSIP Debugging enabled
21:10.52pollerOh yeah
21:10.55pollerI'm game
21:10.58hbsmurfwe've used sip softphones over the internet with it before
21:11.40hbsmurfaccess-list acl-in permit tcp any host 68.76.27.244 eq 5060
21:11.40hbsmurfaccess-list acl-in permit udp any host 68.76.27.244 eq 4569
21:11.40hbsmurfaccess-list acl-in permit udp any host 68.76.27.244 range 10000 20000
21:11.40hbsmurfaccess-list acl-in permit udp any host 68.76.27.244 eq 5060
21:11.44*** join/#asterisk heh_v_water (n=heh_v_wa@71-32-211-123.hlna.qwest.net)
21:11.49hbsmurfSo I've got 5060 tcp and udp open to my asterisk box
21:11.51*** part/#asterisk ahigerd (n=ahigerd@adsl-75-19-79-5.dsl.wchtks.sbcglobal.net)
21:11.54hbsmurf10000 to 20000 open
21:11.56hbsmurfand iax
21:11.57*** part/#asterisk andresmujica (n=AndresMu@190.24.71.182)
21:12.18De_Monack. I just put asterisk into a goto loop
21:12.19hbsmurfopen 5060 tcp
21:12.20*** join/#asterisk champster (n=asterisk@AH.tescogroup.com)
21:12.51hbsmurfI need a voip connection to my favorite pizza place
21:12.57hbsmurfActually
21:13.08hbsmurfThe first part of my home automation project comes in on Tuesday
21:13.11DaeJeon-Newbieanyone able to help me plz
21:13.14hbsmurfGoing to tie it into Asterisk at my house
21:13.21hbsmurfWhat up, newbie?
21:13.44DaeJeon-NewbieI am installing a vxml broswer
21:13.51DaeJeon-Newbiebut
21:13.54DaeJeon-NewbieRegistered application 'Vxml'  == Parsing '/etc/asterisk/vxml.conf': Found    -- Bad Video codec.
21:14.11DaeJeon-Newbiehow can I fix it?
21:14.15Qwell[]DaeJeon-Newbie: I doubt anybody here is going to be able to help you with that
21:14.21hbsmurfAh, vxml
21:14.24champsterWhere do I start to troubleshoot that My 1.2.12.1 PBX will not do any registrations to SIP providors. I can use SIP Outbound, so I do not think it is a firewall issue. I think that asterisk is not even trying to register.  Please advise.
21:14.27hbsmurfNo comprende vxml
21:14.43DaeJeon-Newbiewhere can I get help?
21:14.52hbsmurfWhat is that browser for?
21:14.56pollerb11d: http://tmp.poller.se/sipdebug That aint telling me that much, sorry to say. :(
21:15.16hbsmurfchampster:  Upgrade to 1.2.13 and try again
21:15.23DaeJeon-Newbiehbsmurf . it does TTS AND ASR
21:15.29hbsmurfpoller:  did you open tcp 5060?
21:15.36hbsmurftts and asr?
21:15.39champsterTypo from memory. it is 1.2.13
21:15.44Kattyi do like me some port 5060
21:15.44pollertcp/5060, nope.
21:15.46hbsmurfok
21:15.46Kattyit's hot.
21:15.53hbsmurfopen it
21:15.56pollerWhat's tts and asr?
21:15.59hbsmurfopen tcp 5060
21:16.00pollerhbsmurf: Will do. :)
21:16.20hbsmurfSIP is about as hot as D&D!
21:16.25DaeJeon-Newbietext to speech-tts
21:16.27hbsmurfWOO BABY!  ROLL ME A 12!
21:16.31hbsmurfah, tts
21:16.35hbsmurfasr?
21:16.38hbsmurfI should know this
21:16.44hbsmurfspeech rec?
21:16.45champsterTCP 5060 in addition to UDP 5060?
21:16.47DaeJeon-Newbieasr-automatic speech reco
21:16.48hbsmurfyes
21:16.52hbsmurftcp and udp 5060
21:16.59pollerDidn't change anything.
21:17.02*** join/#asterisk docelm0 (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net)
21:17.05hbsmurfnewbie:  What does the vxml browser have to do with it?
21:17.22hbsmurfpoller:  Well crap
21:17.32hbsmurfpoller:  You have the same ports open that I do
21:17.32pollerI should do some 802.1q tagging and put the asterisk in the dmz to.
21:17.34DaeJeon-Newbievxml broswer is the gateway
21:17.44hbsmurfnewbie:  What package are you using?
21:17.52pollerhbsmurf: You have tcp&udp/5060 and udp/10000-20000?
21:17.59hbsmurfYes
21:18.04*** join/#asterisk Kapsel (i=linknet@irc.thinkgeek.dk)
21:18.04Kattydocelm0: mew.
21:18.08hbsmurfI mean, poller: yes
21:18.15pollerWell, can't it be anything else?
21:18.33DaeJeon-Newbieit receives the call from pstn/sip/gsm , and converts into vxml code
21:18.38champsterNot the TCP, I am adding it now.
21:18.39hbsmurfpoller:  You don't have nat=yes in your sip.conf for your sip trunk, have you tried that?
21:18.45DaeJeon-Newbieand vice versa
21:18.54champsterI have nat=no as I am not using NAT
21:19.10hbsmurfchampster:  Hold on, I've got two sip trunk conversations going
21:19.20hbsmurfchampster: and I'm getting cornfused
21:19.21hbsmurf:)
21:19.27hbsmurfThis is what I have in my firewall
21:19.33hbsmurftcp and udp 5060 open to my Asterisk box
21:19.37pollerhbsmurf: Nope. Should that be under the digisip-9900 and digisip-9901 contexts?
21:19.39hbsmurf10000 to 20000 udp open
21:19.44champsterthx
21:19.46hbsmurfpoller:  Try both
21:19.52pollerUes
21:19.53pollerYes
21:20.02hbsmurfIn my sip.conf I've got nat=yes for anything calling in from the outside
21:20.06hbsmurfJUST IN CASE
21:20.12hbsmurfI mean, I've got nat=yes just in case
21:20.14hbsmurfI hate sip over nat
21:20.23hbsmurfalmost as much as vegetables on pizza
21:20.38pollerStill nothing hbsmurf
21:20.44hbsmurffrickin a
21:21.21poller<PROTECTED>
21:21.42pollerDosn't that mean that asterisk is trying to connect the nokia and my sip-provider directly?
21:22.06hbsmurfThat means it's trying to connect 15799 to your e60
21:22.08b11dbelieve it or not, if done right.. vegetables on pizza can be good.. but its very rarely accomplished
21:22.10hbsmurfor vice versa
21:22.38pollerhbsmurf: In my case, isn't that a bad thing?
21:22.40hbsmurfWell, not directly
21:22.52hbsmurfyou have reinvite=no
21:22.57hbsmurfwhich means Asterisk will bridge the two calls
21:23.03hbsmurfbut still
21:23.08hbsmurfthat looks right to me
21:23.15hbsmurfyou're just not getting two way audio
21:23.15hbsmurfright?
21:23.19b11dRTP..
21:23.22pollerright
21:23.34b11dI had a sick RTP issue.. took a week to figure out..
21:23.34hbsmurfWhich way?
21:23.35filecanreinvite=no
21:23.38hbsmurfWhich end?
21:23.38fileis the correct option
21:23.40b11dstupid routing error
21:23.44hbsmurfcanreinvite, whatever
21:23.46pollerhbsmurf: The outside party can hear me.
21:23.53hbsmurfbut you can't hear them?
21:23.55hbsmurfso it's incoming rtp
21:23.59pollerhbsmurf: Yepp.
21:24.09hbsmurfWhat firewall?
21:24.22pollerFreeBSD/PF
21:24.29hbsmurfYou don't have an outside address in your sip.conf either, do you?
21:24.37pollerNo.
21:24.42poller*feeling stupid*
21:24.43poller:)
21:24.48b11dPF == greatest firewall ever
21:24.51hbsmurfgimme  asec
21:24.59pollerhttp://tmp.poller.se/sip.conf
21:25.02pollerThanks :)
21:25.04pollerb11d: <3
21:25.13b11d:)
21:25.14champsterIs there a way to force asterisk to register with a providor? sip reload isn't doing it. My sip show registry is just blank.  I can make SIP call to those providors just fine, so I do think it is a FW issue.
21:25.21pollerBut, i'm not doing any firewalling with mine. :)
21:25.21hbsmurfexternhost
21:25.30hbsmurfno no no
21:25.31hbsmurfexternip
21:25.36pollerhbsmurf: Where's that going?
21:25.43poller[general]?
21:25.49hbsmurfhttp://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip
21:25.50hbsmurfYes
21:25.57hbsmurfThat's why
21:26.03hbsmurfyour voip provider doesn't know how to get back to you
21:26.09hbsmurfit's lost and can't find the way
21:26.11hbsmurfprobably crying
21:26.13hbsmurfyou mean mean person!
21:26.31hbsmurfSee what you've done?
21:26.34b11d:)
21:26.38poller:(
21:26.39hbsmurf:)
21:27.04pollerStill not working.
21:27.04poller:(
21:27.12hbsmurfHold on a sec partner
21:27.18pollerRoger that. :)
21:27.22hbsmurfYou put in externip with the outside address of your firewall
21:27.29hbsmurfor at least the outside address of your asterisk box
21:27.31pollerOfcourse. :)
21:27.37hbsmurfyou put nat=yes in the sip.conf for those two connections?
21:27.40b11dugh..  63 more minutes..
21:27.42pollerYepp
21:27.48hbsmurfhold on
21:27.52pollerThanks
21:28.28hmmhesaysso anyone speak german in here?
21:28.43champsternine
21:28.45filehmmhesays: nein
21:28.49pollerBitte!
21:28.51champsteroops
21:29.05hbsmurfGerman?
21:29.07pollerchampster: Good try thou
21:29.14champsterdas ist goot
21:29.17hbsmurfI drive a VW!
21:29.31pollerchampster: Or 'gutt', i don't know. ;)
21:29.34hbsmurfOk
21:29.36champsterlol
21:29.37hbsmurfWhat I've found is this
21:29.44hbsmurfYou need nat=yes in the numbers
21:29.46hbsmurfand externip in general
21:29.49champsteronly light conversational
21:29.55b11dwhat is that saying germans say.. sounds like "brussush los"
21:29.56pollerhbsmurf: Check
21:29.57b11dor something like that
21:29.58champsternot written apparently
21:29.59hbsmurfand your register statements are right
21:30.11hbsmurfAre you using the correct codec?
21:30.14hbsmurfCould that be it?
21:30.24hbsmurfput nat=no on your e60 entry too
21:30.26pollerWhy not? I have no idea. :)
21:30.29pollerOk
21:30.32hbsmurfHave you restarted or just reloaded?
21:30.47pollerrestart
21:30.50hbsmurfok
21:30.51pollerJust to be safe
21:30.51poller:)
21:30.53hbsmurfIt looks right o me
21:30.54hbsmurfto me
21:31.00hbsmurfI'm looking at this:
21:31.04hbsmurfhttp://www.voip-info.org/wiki/view/Asterisk+FWD+NAT+Config+Example
21:31.14pollerCan it be the codecs?
21:31.17hbsmurfCould be
21:31.21hbsmurfWhat codecs are you using?
21:31.27polleralaw and ulaw
21:31.33pollerHave no idea if thats good.
21:31.50b11dg711 baby
21:31.51b11dWOOH
21:31.52hbsmurfwhere are you located?
21:31.58pollerSweden
21:32.00hbsmurf80KBPS OF VOIP POWER!
21:32.03hbsmurfSweden?  Sweet!
21:32.03Kattysweden is awesome.
21:32.09hbsmurfEver heard of The Flower Kings?
21:32.17pollerNo. :)
21:32.19hbsmurfWhat codecs does your provider support?
21:32.20hbsmurfDamn!
21:32.20pollerCan't say i have.
21:32.26hbsmurfGreat Swedish prog band
21:32.29Kattypoller: will you hide me in your closet?
21:32.33hbsmurfSaw them in Chicago a month or two ago
21:32.34pollerKatty: Any day
21:32.36hbsmurfAMAZING concert!
21:32.36Kattypoller: 'merica isn't so much fun.
21:32.42*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
21:32.48*** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
21:32.55hbsmurfAmerica is fun, as long as you stay away from the whackos.  :)
21:33.01hbsmurfNow Canadia scares me!
21:33.11hbsmurfOkay, back on task
21:33.22hbsmurfWhat does your provider want?
21:33.25hbsmurfI'm assuming alaw
21:33.28Kattycookies and milk.
21:33.30hbsmurfor gsm or something
21:33.33Kattyi'd want cookies and milk.
21:33.34hbsmurfg.729?
21:33.35monstedthere are non-whackos in the US?
21:33.37hbsmurfI've got cookies
21:33.38hbsmurfYes
21:33.41hbsmurfI'm not that whacko
21:33.45hbsmurfat least not much
21:33.47Kattyi am.
21:33.50hbsmurfbut that's another story
21:34.13linageerobl^: http://www.poppymom.com/archives/2006/03/the_naughty_nun.php
21:34.20pollerhbsmurf: I don't know.
21:34.21hbsmurfI need to go to Sweden to see The Flower Kings and the other bands that are spawned from them
21:34.22hbsmurfgood times
21:34.30hbsmurfpoller:  What itsp are you using?
21:34.35pollerdigisip
21:35.13hbsmurfWell then
21:35.21pollerWell what? :)
21:35.26hbsmurfI can't read it because I'm sheltered
21:35.27hbsmurfGimme a sec
21:35.29hbsmurf:)
21:35.56pollerDigisip stödjer ett flertal codecs. g711ulaw, g723r63, g723r53 och g729r8. Standard-codec som skall användas är G.711. Om kvalitén blir för dålig med G711 kan du testa G.729.
21:35.58pollerSorry for swedish. :)
21:36.03hbsmurfNot your fault
21:36.03hbsmurf:)
21:36.06pollerThat's from their website. :)
21:36.15hbsmurfAh, is alaw or ulaw first in your list?
21:36.18hbsmurfmake sure ulaw is
21:36.25polleralaw is
21:36.26poller=b
21:36.27hbsmurftake out alaw
21:36.37rudholmor even µlaw :)
21:36.41hbsmurfdude
21:36.47hbsmurfDUDE THAT FREAKS ME OUT
21:36.47hbsmurfheh
21:36.59hbsmurfHi1
21:37.00rudholmit'll be ok.
21:37.02hbsmurfwrong window!
21:37.03hbsmurfdammit
21:37.13hbsmurfOk
21:37.17hbsmurfAny luck?
21:37.20pollerhbsmurf: Same problem with no alaw.
21:37.31hbsmurfdammit
21:37.39hbsmurfAre you SURE your firewall is right?
21:37.42pollerMgm :(
21:37.44hbsmurfI mean, freebsd and all that
21:37.46hbsmurf:)
21:37.46pollerYou can't be SURE
21:37.49poller:)
21:37.51hbsmurfyou're right
21:37.55hbsmurfHmmm
21:38.18pollerGah, i've been working since 6 this morning, i can't think.
21:38.26pollerStarting sniffing packas will have to wait. :)
21:38.27hbsmurfIt HAS to be a firewall issue
21:38.31hbsmurfheh
21:38.36pollerBAD BAD BAD FIREWALL! :(
21:38.39hbsmurfI'm running a Cisco PIX firewall
21:38.47poller*bah bah*
21:38.47poller:)
21:38.47hbsmurfit works with asterisk
21:39.02b11djust make sure you have specific entries to handle the ROUTEs
21:39.04pollerI'm going to build some vlans now instead.
21:39.06b11dI had to do that to fix my RTP issue
21:39.08hbsmurfon the pix?
21:39.09b11dwith one way audio
21:39.10hbsmurfthe pix doesn't route!
21:39.14pollerPuting asterisk in both dmz and internal.
21:39.15b11dsigh
21:39.18b11dyou still need to set the routes
21:39.18hbsmurf:)
21:39.24hbsmurfstatics and access lists
21:39.24b11dI had to do it on my VG-224.
21:39.24linageehbsmurf: did you click the link?
21:39.24hbsmurf:)
21:39.33hbsmurfhold on
21:39.37b11dyou could ping off that vg224 anywhere..  everything worked.
21:39.38b11dexcept RTP
21:39.38pollerb11d: What entries?
21:39.40hbsmurfthe vg-224 was an amazing piece of hardware
21:39.50b11duntil I added a "ip route x.x.x.x/x.x.x.x x.x.x.x" entry
21:39.51hbsmurfwere you doing h.323?
21:39.56b11dno.. sip
21:39.56hbsmurfoh, THAT route
21:40.06b11dI *had* to set one..
21:40.16b11deven though all other IP traffic seemed to be functioning
21:40.19hbsmurflinagee:  That's jsut wrong
21:40.20hbsmurf:)
21:40.25b11dyeah.. it bothers me as well
21:40.31hbsmurfCisco devices need routing info
21:40.35hbsmurfI'm a Cisco whore
21:40.38b11dsame
21:40.40pollerMy avaya is doing h.323 like CRAZY! :)
21:40.42hbsmurfUsed to have a whole bunch of certs
21:40.43b11dCisco owns me
21:40.48hbsmurfI let them expire
21:40.50b11dsame
21:40.52pollerAvaya owns cisco. :)
21:40.54hbsmurfI need to start testing again
21:40.55b11dI used to be CCIE
21:40.56hbsmurfDUDE
21:40.57b11dok thats a lie..
21:40.58hbsmurfAVAYA?
21:40.58champsterI figured it out my register prob.  There was a device being read in before my registers, so they were no longer in [general] context
21:40.59hbsmurfNo
21:41.01hbsmurfI was ccnp ccdp
21:41.04champsterthanks
21:41.07hbsmurfand voip specialized
21:41.16b11dthat'll happen champster
21:41.17hbsmurfchampster:  Ah!  good for you!
21:41.23*** join/#asterisk swilliamson (i=swilliam@209.42.110.46)
21:41.28hbsmurfNow I'm just ccna
21:41.29pollerhbsmurf: I'm working with avaya.
21:41.33hbsmurfI need to take my ccda
21:41.36champsterguuten tag
21:41.40pollerDon't know why realy, just happend. :)
21:41.52*** part/#asterisk ManxPower (n=manxpowe@231.sub-75-201-47.myvzw.com)
21:41.53b11dwtf is that "russhosh lohss" the germans say..
21:41.55hbsmurfpoller:  I only touch the Avaya Partner or Merlin stuff, none of the voip equipment
21:41.55swilliamsonany of you guys working on the realtime ldap stuff
21:41.56champsteraufvieterzein
21:42.03hbsmurfrealtime ldap?
21:42.09champsterlol
21:42.18cthorneryeah
21:42.21hbsmurfrealtime ldap sounds like it's another layer of frustration on top of realtime
21:42.25pollerhbsmurf: It was like: Oh, the phone-guy wants to quit.. HERE POLLER! TAKE THE PBX! IT'S YOURS!
21:42.27hbsmurfwhy wouldn't you just use realtime without ldap?
21:42.35*** join/#asterisk _spirit_ (n=spirit@66.161.100.230)
21:42.41hbsmurfpoller:  Been there, done that.  It's how I got into Cisco and voice stuff.  :)
21:42.43cthornerI'm working on realtime ldap
21:42.52hbsmurfWhat are you using for an ldap back end?
21:42.58cthornereDir
21:42.58hbsmurfAnd what are you storing in it?
21:43.00b11deDirectory :)
21:43.03b11dHAHAHAHA
21:43.04b11dREALLY?
21:43.07swilliamsonldap back end? data source is edirectory
21:43.09cthornerreally
21:43.10hbsmurfI'm sorry, I can't talk to you anymore
21:43.11hbsmurf:)
21:43.13pollerhbsmurf: We use mostly digital telephone but have som IP-phones. Not SIP thou.
21:43.14b11dhahahahahaha.. man that sucks :)
21:43.26hbsmurfI'm in the middle of a Groupwise to Exchange migration
21:43.29b11dshudder
21:43.31cthornerok
21:43.31b11dI like groupwise
21:43.32hbsmurfI can't wait to shut that damn Groupwise server off
21:43.33hbsmurf:)
21:43.36hbsmurfNo no
21:43.37hbsmurfReally
21:43.39b11dI have no issue with GW at all
21:43.39hbsmurfeDir isn't bad
21:43.40pollerExchange > groupwise
21:43.41swilliamsoni am using our ldap directory as the source for email addresses and stuff
21:43.58hbsmurfWhat else would you pull via ldap for Asterisk though?
21:44.07swilliamsonso there is no more provisioning of voicemail infos
21:44.07cthorneruser info
21:44.14hbsmurffor endpoints?
21:44.15swilliamsonsip accounts, configs
21:44.15monstedexchange is crap, but unfortunately the only choice :(
21:44.20hbsmurfvia ldap?
21:44.20cthornermost big companies have an extensive user tree
21:44.22hbsmurfInteresting
21:44.25b11dexchange has "features" but big deal..
21:44.27swilliamsondialplan
21:44.33SplasPooddamn, seems my connection keeps getting dropped trying to download asterisk-sounds from ftp.digium from China
21:44.33b11dim looking to see where this "eGroupware" project goes..
21:44.33hbsmurfDIALPLAN FROM LDAP?
21:44.34swilliamsonyou can get everything
21:44.35cthorneryeah,
21:44.37hbsmurfWhy don't you just start drinking now
21:44.53hbsmurfI admit
21:44.54alamantiahowdy cthorner
21:44.56hbsmurfIt sounds intresting
21:44.58hbsmurfinteresting too
21:45.06Sed[PCT]hbsmurf: with regards to groupwise... you don't work for a affiliate school of Penn State, do you?
21:45.08cthornerhi, that's a better nick
21:45.10hbsmurfI guess I don't have any clients big enough to care about it though
21:45.15hbsmurfsed: no
21:45.24alamantiayup
21:45.25hbsmurfsed:  Migrating a county government in northern michigan
21:45.28cthorner\msg alamantia hey
21:45.31hbsmurfsed: they should've never been on groupwise
21:45.31alamantia:)
21:45.37cthornerdidn't work did it
21:45.37hbsmurfsed:  it's a long story
21:45.40b11dremember.. there was a time when you DID run Novell products..
21:45.41swilliamsonso cthorner, what have you done to your edirectory schema? have you modded it at all
21:45.42alamantiait's the wrong \
21:45.44b11dNovell used to ROCK
21:45.46hbsmurfI started out on Novell
21:45.47hbsmurf3.11
21:45.50b11dso.. so many people stuck with it..
21:45.50hbsmurfIt owned
21:45.52b11dthats why GW is everywhere
21:45.57hbsmurfYeah
21:46.06hbsmurfI just installed a two node 6.5 cluster for a school in August
21:46.07hbsmurfit's awesome
21:46.08b11dfirst thign I did when I took over here.. got rid of ALL Novell products
21:46.14b11dexcept GroupWise (and eDir)
21:46.17hbsmurftoo bad his nds is flaky as hell
21:46.21b11d6.5... why not 7?
21:46.25Sed[PCT]hbsmurf: as, ok.. see.. I personally love novell... but my college is switching from GW to Exchange (and from novell to microsoft in general) over the next month (servers anyway, clients are done)
21:46.30hbsmurfthey own 6.5?  :)
21:46.34b11doh ;)
21:47.02b11dthe state of MN is still all in love with Novell..
21:47.05hbsmurfsed:  I've been doing Microsoft stuff since I moved to Northern Michigan, so about 10 years now. I've also done Novell in taht time but there's so much more I can do with Microsoft that I don't bother with Novell
21:47.05b11di hate them
21:47.08b11dand their "partners"
21:47.09*** join/#asterisk DaeJeon-Newbie (i=Singh@124.62.150.38)
21:47.13hbsmurfState of MI is moving to Microsoft now
21:47.28Sed[PCT]hbsmurf: yes... and I can say the same with novell over microsoft.. it just depends which you know better...
21:47.29hbsmurfI just hate the fact that I've got to run Zen to do anything decent as far as policies go
21:47.31b11di went to a samba backend..
21:47.43DaeJeon-NewbieQwell:I solved the problem
21:47.48hbsmurfsed:  I know both fairly well, I just like not having to run seperate packages to manage my PCs.  :)
21:47.55hbsmurfSamba is interesting
21:48.03Sed[PCT]heh
21:48.03b11dsamba4 looks really cool
21:48.03b11dwe'll see though
21:48.07hbsmurfI installed a 1.7TB samba server for a client
21:48.13hbsmurfRight when 3.0 came out
21:48.17b11dthats about a third of what im handling
21:48.19hbsmurfThe first couple months were not very smooth
21:48.20DaeJeon-Newbie<PROTECTED>
21:48.34hbsmurfYou must remember, I'm in Northern Michigan
21:48.37hbsmurfIn a tourist area
21:48.44b11dmy only issue is that every now and again.. some PC's "fall off the domain"
21:48.46hbsmurfI left a life a huge networks behind 4 years ago
21:48.46hbsmurf:)
21:48.47b11dits annoying
21:48.54b11dhell.. im in Northern Minnesota!
21:48.55hbsmurfYeah, I've had that with Samba
21:49.02b11d18k people here..
21:49.02b11dthats it
21:49.09hbsmurfbut you're at an institute of higher learning, right?  :)
21:49.16b11dyes..
21:49.16Sed[PCT]b11d: thats a microsoft feature
21:49.16b11d"higher" is loosely defined
21:49.17*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
21:49.18b11dhaha sed..
21:49.20hbsmurfWe sell equipment to our local college but I don't get into their network
21:49.27Sed[PCT]the k-12 district I help  with over breaks.. had that happen a lot...
21:49.29swilliamsonsame here, canadian university in ontario
21:49.32hbsmurfThey've got a 4 node Netware cluster with fiber channel back end
21:49.32b11dwhat one?
21:49.34hbsmurfit's pretty sweet
21:49.35Sed[PCT]and my college.. when we were migrating.. it was happening
21:49.37b11dI went to UWO
21:49.45swilliamsontrent
21:49.47b11dcool
21:49.53b11dUWO > *
21:49.55b11d:)
21:49.57swilliamsonha
21:50.06swilliamsonI did one year there, then transfered
21:50.07b11dUBC is aweomse though
21:50.14swilliamsonliked it though
21:50.18b11dI really enjoyed London.. period.
21:50.23swilliamsonwill always love london
21:50.27*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
21:50.32hbsmurfok
21:50.37hbsmurfThe phoen hasn't rang in 30 minutes
21:50.42hbsmurfScrew this, I'm leaving
21:50.45b11dhehe.. just stay north of southdale :)
21:50.46hbsmurfI own the place and I'm the last one here!
21:50.48hbsmurfwtf!
21:50.54b11dfire everyone.. hire us..
21:51.16hbsmurfDo you work cheap?
21:51.18hbsmurf:)
21:51.22b11ddope and beer is all I need
21:51.31pollerI have 42 degrees C in my closet.
21:51.33poller:(
21:51.34b11ddope meaning weed
21:51.38b11dnot hard drugs :)
21:51.45b11dthats far, far too hot
21:52.03pollerMmm. :(
21:52.22hbsmurfDamn, the phone just rang
21:52.22pollerI can dry clothes in there. :)
21:52.30hbsmurfair conditioning ftw!
21:52.41hbsmurfI wish we had some snow
21:52.45hbsmurfthen it wouldn't be so damn hot in my office
21:52.53hbsmurfnorthern michigan and there's no snow in December
21:53.02hbsmurfel nino pisses me off
21:53.08b11dihear you
21:53.10*** join/#asterisk Winkie (n=urmom@86.149.146.70)
21:53.27hbsmurfok
21:53.38hbsmurfI'm going home to build three Asterisk servers and figure this realtime crap out
21:53.41b11dno
21:53.42b11dyour not
21:53.45hbsmurftalk about an exciting friday night!
21:53.49b11dgo home.. get drunk.. smoke a fatty.. and get laid
21:53.50hbsmurfOk, I'm not
21:53.51swilliamsonha, realtime is fun
21:53.55hbsmurfI'm going to go do my wife
21:54.00hbsmurfI have no dope
21:54.03hbsmurfMaybe my neighbors do!
21:54.03b11dok :)
21:54.06b11dhahaha
21:54.06hbsmurf:)
21:54.26b11dyou have a good weekend..
21:54.26swilliamsonI am out, just wanted to connect with the ldap guys. see ya all later
21:54.27hbsmurfI was supposed to go see a client this afternoon and I'm POSITIVE the manager is stoned all the time
21:54.30hbsmurfShe just looks like it
21:54.31b11dtalk to you later swilliamson..
21:54.32b11dUWO RULES
21:54.32hbsmurfThanks, you too!
21:54.33b11d:P
21:54.38hbsmurflater!
21:54.44swilliamsonsee ya b11d
21:54.44pifiulolol
21:54.46b11dhehe
21:54.46*** part/#asterisk swilliamson (i=swilliam@209.42.110.46)
21:55.01b11dfinally..
21:55.05pifiuim bored
21:55.06b11dwe can get down to some serious talk now..
21:55.07b11d:)
21:55.08pifiucounting down the hour
21:55.12b11dyeah.. 35 more mins here
21:55.28pifiui have shit to do but im burned out, need a break badly
21:55.38b11dgo do something that's interesting
21:55.45De_MonGaaah
21:55.45b11dor go for a 20 minute walk or something
21:55.53*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
21:55.59De_MonRetryDial's Goto option appears to be broken
21:56.25b11dcheck the bug reports
21:56.26b11d?
21:58.11b11dim going to watch so much god damn Star Trek this weekend..
21:58.16b11dgod im a loser..
21:58.20b11di need to take my own advice..
21:59.55DaeJeon-Newbieplease help me anyone
22:00.15b11duhh.. give me a Z..
22:00.17b11da Q
22:00.19b11danother Q
22:00.20DaeJeon-Newbiei recently installed the asterisk-gui
22:00.22b11da third Q
22:00.24DaeJeon-Newbie== Parsing '/etc/asterisk/manager.conf': Found
22:00.24b11dthe number 4
22:00.25DaeJeon-Newbie<PROTECTED>
22:00.28b11dand the batman symbol
22:00.42DaeJeon-Newbiewhat is the problem?
22:00.59b11ddunno..
22:01.30DaeJeon-NewbieQwell:?
22:01.43DaeJeon-Newbieit is about gui
22:02.32b11dgod this bothers me:  http://sportsillustrated.cnn.com/multimedia/photo_gallery/0612/gallery.cowboys.stadium/content.1.html?cnn=yes
22:06.10DaeJeon-Newbieplease help me anyone
22:07.18[TK]D-FenderDaeJeon-Newbie` : please read the channel topic
22:07.57DaeJeon-Newbieyes sir
22:10.51pollerYES
22:10.56pollerb11d: It's working.
22:11.02pollerIt was the nat. =b
22:11.12pollerLucky me, having plenty if ip's :)
22:11.19De_Monhttp://pastebin.ca/280408 my bug is that G() doesn't work correctly in retrydial...
22:12.13b11d:)
22:12.15mercestespoller:  Or you could do your natting correctly
22:12.16b11dcool.. im glad to hear its fixed
22:12.39b11dftw!
22:12.43pollermercestes: Either that or it was something bad with the config.
22:12.57mercestesthere are some sip.conf setings.
22:13.00mercestesexternip and crap like that.
22:13.03pollerNow i have another problem. :) I can't dial out. :)
22:13.09pollerDec 15 23:12:02 WARNING[3506]: chan_sip.c:9720 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"e60" <sip:e60@213.112.11.41>;tag=as7e106a37'
22:13.20mercestesNice..what's your password?
22:13.33pollerI'm not telling you. ;)
22:13.36b11dlol
22:13.43mercestesit looks like your password is e60
22:13.48mercestesand your username is sip
22:13.50b11dlol
22:14.00pollerThats bad.
22:14.06*** join/#asterisk tuck3r_ (n=tuck3r@unaffiliated/tuck3r)
22:14.13poller;)
22:14.19mercestesso I would either change your password.
22:14.24pollerlol
22:14.27mercestesor I would try username:password@ipaddress.
22:14.28pollerGood solution
22:14.29poller:)
22:14.33mercestesinstead of sip:username@ipaddress
22:14.57pollerI love those ugly solutions. :)
22:15.16b11dohhh.. i so want one of these:  http://news.xinhuanet.com/english/2006-12/14/content_5487567.htm
22:15.23mercestesI love it when someone msg nickserv identify anatomatopea
22:15.51b11dhaha
22:15.55b11ddamn nickserv..
22:16.17b11di still think irq's "hot ass" website should be in the topic
22:16.29*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
22:16.44mercestesman, someone yelled at me about being off topic for asking questions on a polycom once.
22:16.51b11dhaha
22:16.52b11dand here I sit
22:16.53b11dall day
22:16.55b11doff topic
22:16.55b11d:)
22:16.58*** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.wa.comcast.net)
22:17.09mercestesYea, I told him to shut his damn pie hole.
22:17.12b11dlol
22:17.13mercestes...and then he banned me.
22:17.17b11dthat'll happen
22:17.20mercestesdidn' tknow he was +o. lol
22:17.23b11dthe people here are pretty cool though
22:17.40mercestes*shrugs*  Actually he didn't ban me, he just kicked me.
22:17.44mercestesso all things considered, yea, pretty cool
22:17.45b11dstill..
22:17.59pollermercestes: My nokia adds sip: in front of the "public user name" when i try to edit.
22:17.59mercestes*shrugs*  I figured it was a time out..lol
22:18.00b11dim pasting the link..  cant let this one die
22:18.01b11dhttp://zeppelin.stepahead.net/~dan/pz/list.html
22:18.06irqheh
22:18.10b11dhahah you're still here
22:18.15irqalways glad to help
22:18.30b11dspend your entire weekend browsing usenet groups to add to that collection
22:18.30irqmaybe we should put like [NSFW] on it
22:18.36b11dspend no time on other things :)
22:18.44mercestesomg, what is that?
22:18.44rudholmthat depends on where you work
22:18.46irqi just found them all at once
22:18.55pollermercestes: Shouldn't this work, i mean, my settings works to register with.
22:19.24Kattythere anything niftier than iaxcomm?
22:19.34irqb11d: where you from?
22:19.39mercestespoller:  Depends on if it's really sending sip:e60@ip.
22:19.45mercestesb11d:  what's the point of that hot website?
22:19.48b11dOntario, Canada originally.. i live in Northern Minnesota now..
22:19.52pollerThat ip in the error message is my asterisks external ip.
22:19.53b11dthe point is ass..
22:19.54b11dthats it..
22:20.06mercestesah...it is a nice ass
22:20.12b11dindeed
22:20.14mercestesis there any sans pants?
22:20.18b11di like #50 myself
22:20.30b11dbut I havent seen all yet
22:20.30b11d:P
22:20.32mercestesis she single?
22:20.40b11dprobably..
22:20.44b11dor at least.. doesnt care..
22:20.45b11d:)
22:21.12Kattyi'll take that as a no.
22:21.12b11dnah.. thats probably wrong of me to say..
22:21.14irqmercestes: yes, there is. sans thong even, a few times
22:21.23mercestesirq:  really?  Where?
22:21.34irqjust keep scrolling through thte page, you'll find them
22:21.44mercestesb11d: Maybe she does care...and she's nto single...but her boyfriend makes her do it and she complies out of her undying love for him.
22:21.46mercestesthat's hot.
22:21.54*** join/#asterisk c4t3l (n=c4t3l@69.15.174.114)
22:22.03mercestesc4t3l!  What'd you break now you tard?
22:22.34b11dlol
22:22.42pollerOk, i want to call out. :)
22:22.43b11dnice mercestes
22:23.06mercesteshe only comes in here when he breaks something and wants to know how to fix it..:D
22:23.13b11dwhat a bastard..
22:23.16mercestesYea...
22:23.17b11dat least crack a joke or something
22:23.49b11dor post a link to some nice ass
22:25.54poller:)
22:26.33mercestesI want some hot ass....
22:26.40irqwe just url'd you on some
22:26.42mercestes....I miss my wife...I wanna go home.
22:26.43b11din order to get some, you need to get off of the IRC
22:27.14linageeshellshark: i can't find on your site, what is the iax server name? i want to check reliability before i sign up
22:27.16*** part/#asterisk cthorner (n=cthorner@209-234-185-148.static.twtelecom.net)
22:27.21*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)
22:27.27b11dwell gentlepeople..
22:27.31b11dtime for me to go home
22:27.35b11dhave a great weekend chaps
22:27.37irqcya :)
22:27.38irqyou too
22:27.43linageeb11d|bbl: later
22:28.17linageeack. i want to do something cool with my voip phone display. :)
22:28.49linageelike maybe display my email count or something
22:29.22robl^linagee: XML apps are kewll.. like with Cisco or Aastra 480is ;_
22:29.33*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)
22:30.01linageerobl^: even displaying the day of the week would be nice. (phone only displays the numeric)
22:30.24robl^linagee: what phone?  
22:30.33linageerobl^: it would be really neat to display something like web hits
22:30.37linageerobl^: gxp-2000
22:30.47linageerobl^: i know there's probably a way
22:30.52robl^eww
22:30.55linagee:p
22:31.22robl^only thing I like about those phones are the massive BLF expansion module.. lots of buttons and lights
22:31.33linageerobl^: yes
22:31.42mercestesHey, Fender, aren't you the polycom deity?
22:31.50linageerobl^: i have seen phones in TV series that are color screen. that almost seems silly. lol
22:32.21robl^Some Cisco phones have color screens...
22:32.21linageeyeah, and why not have a keyboard and mouse on the phone too! lol
22:32.29mercestesanybody know how to disable the PC port on a Polycom [5..6]01 phone?
22:32.39irqi've had a lot of luck with the low-priced gxp-2000
22:32.45*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
22:33.27[TK]D-Fendermercestes : Questions are free, answers are $4.95/m ;)
22:33.46*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.14, 1.4.0-beta4, Zaptel 1.2.12, 1.4.0-beta3 Released! (Dec. 15, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework
22:33.53linagee[TK]D-Fender: just call 1-900- and reference your case number. :)
22:34.34linageemercestes: PC port? politically correct?
22:34.45mercestes[TK]D-Fender:  Do you kno whow to disable the PC port?
22:35.06linageemercestes: are you talking about an ethernet "pass through" port? why disable it?
22:35.16mercesteslinagee:  Because I suck.
22:35.20linagee?
22:35.29linageemercestes: really? come with me! :-D
22:35.35mercesteslinagee:  muhahaha.
22:36.26[TK]D-Fendermercestes : Wire cutters.
22:37.02linageemercestes: super glue and a spare RJ-45 punch
22:37.04mercestes[TK]D-Fender:  you know.....I actually thought of that.
22:37.18mercestesExcept I am a little mroe technical...desoldering the leads from the board.
22:37.29linageemercestes: at least with super glue if you really really wanted to use it again, you could probably pry it loose
22:37.43*** join/#asterisk sloth (n=sloth@cpe-69-203-212-182.nyc.res.rr.com)
22:37.45Supaplexhow can I fetch all database keys in a family via agi?
22:37.57linageein a family?
22:38.17Supaplexyea, Usage: DATABASE GET <family> <key>
22:38.23linageeis that asterisk's version of a table?
22:38.59*** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com)
22:39.03SupaplexI guess :-/
22:40.06fileblitzrage: don't jynx it all
22:41.23*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:42.49*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:42.57*** join/#asterisk irq_ (n=dan@wsip-70-168-52-194.sd.sd.cox.net)
22:44.13[TK]D-Fendermercestes : Anyone who realyl wants to latch onto your network will put a switchin front of the phone anyways.  Damned if you do......
22:44.34*** join/#asterisk osas (n=nnnosas@CABLE-72-53-75-244.cia.com)
22:45.19Kattynight guys
22:48.03*** join/#asterisk hbsmurf (n=ghandi@68-188-139-162.dhcp.aldl.mi.charter.com)
22:48.30hbsmurfok
22:48.40hbsmurfhow many more 1.4 betas do you think there will be?>
22:50.23De_Monwell that would depend on the number of bug reports
22:50.33De_Monstop reporting bugs and it will leave beta :P
22:50.39hbsmurfThere's always someone that wants to inject reason into a discussion
22:50.40hbsmurf:)
22:50.42russellbhonestly, this will likely be the last one
22:50.44hbsmurfI'm not reporting bugs!
22:50.55hbsmurfI'm actually thinking of switching over to it this weekend
22:51.07hbsmurfI'd like to get my server at work over to the poundkey dist and off fc3
22:51.13hbsmurfmaybe conary will make updates easier
22:51.19De_MonIm really excited about chanspy whisper mode
22:51.53hbsmurfThat's where someone can whisper something into your ear like a call announcement?
22:52.18De_Monits where the person spying can talk to one channel of the bridged call
22:52.23hbsmurfYeah
22:52.25hbsmurfthat's neat
22:52.36hbsmurfit'd be perfect for announcing important calls to someoen on the phone
22:52.39russellb1.4 also has 120% more sillyness
22:52.42hbsmurfsweet!
22:52.48hbsmurfIt's about time we got sillyness!
22:52.50fileand 42% more muffins
22:52.52hbsmurfI need to learn to program
22:52.56hbsmurfdamn, muffins are fatening
22:53.02hbsmurfcan we have fewer muffins?
22:53.12fileit's a compile-time option
22:53.12hbsmurfWe need a Polycom config generator built into the gui
22:53.13De_Moncall me when it has cookies and cream
22:53.17hbsmurfsweet
22:53.18file./configure --muffin-count=1
22:53.23hbsmurfjust 1?
22:53.31hbsmurfCan we make it so it spits out 1 muffin a day?
22:53.31filewell you said you want fewer muffins
22:53.37hbsmurfLike, for breakfast?
22:53.45filesure
22:53.45hbsmurfI just don't want so many at one time
22:53.50hbsmurfsweet
22:53.53[TK]D-Fenderhbsmurf : Just use Trixbox if you want a system to do your thinking for you...
22:53.59lters_How do folks hand vm to pagers...
22:54.01hbsmurfDude
22:54.03hbsmurfDUDE
22:54.09hbsmurfI don't want it to do my thinking for me
22:54.15hbsmurfI just want a Polycom config generator
22:54.15hbsmurf:)
22:54.20hbsmurfActually, I found one
22:54.24hbsmurfhaven't tested it yet
22:54.25file[TK]D-Fender: I just want
22:54.33lters_is there a way to dial a normal pager num, and drop the callerid digits..
22:54.33hbsmurftrixbox
22:54.35hbsmurfmy goodness
22:54.41hbsmurfdrop?
22:54.45[TK]D-Fenderhbsmurf : You're looking for GUI's that even configure your phones for you, so YES, you most certainly are lumping yourself onto that category.
22:54.59[TK]D-Fenderlooks like a duck, walks like a duck, quacks like a duck....
22:55.03hbsmurfHeh
22:55.04[TK]D-Fenderfile : ! ! !
22:55.06hbsmurfGrouchy?
22:55.07hbsmurf:)
22:55.16hbsmurfSo far I've done all my configuration by hand
22:55.25lters_vi works fine ;)
22:55.30hbsmurfvi is teh devil!
22:55.35hbsmurfI like nano
22:55.37hbsmurfit's easy
22:55.37hbsmurf:)
22:56.10russellbwe have a gui in the works now :)
22:56.11bkruseomg
22:56.14russellbi really like it
22:56.15hbsmurfI'm installing 1.4 beta 3 on a machine next to me and the monitor is so screwed up I can't really see what is happening
22:56.16mercestesvi is klingon for "edit"
22:56.17hbsmurfI love this montior
22:56.40hbsmurfI'd like a gui just for my clients
22:57.03hbsmurfI have a couple that would like to be able to manage the system themselves
22:57.12mercesteshbsmurf:  Operator Flash Panel.
22:57.16hbsmurfAlready using it
22:57.21hbsmurfI'm talking config
22:57.24hbsmurfextension stuff
22:57.38bkrusemercestes: web interface in beta3.
22:57.48hbsmurfOne of my local clients is test driving a Shoretel system
22:57.51hbsmurfthe gui is pretty sweet
22:58.07hbsmurfthey're not the type that would do well with the text file configs in Asterisk
22:58.07bkruseya, were doing so much work on it to
22:58.20hbsmurfmy 1.4 install is almost done
22:58.22hbsmurfpizza just got here
22:58.41hbsmurfPersonally, I like working on the text files
22:58.49bkrusesame
22:58.49hbsmurfit's one less level of stuff to screw up
22:59.07hbsmurfbut the one bank testing Shoretel would love a gui
22:59.19hbsmurfif for nothing other than call routing and checking endpoints
22:59.40hbsmurfOut of all my clients, they're the only one that I would need that for
22:59.48bkruseya, i agree, it gets the basic job done.
22:59.51hbsmurfyeah
22:59.51*** join/#asterisk DaeJeon-Newbie (i=Singh@124.62.150.38)
23:00.10hbsmurfNow if I could get the Polycom phones to do what I wanted...
23:00.21DaeJeon-Newbieasterisk server is running on boot
23:00.27hbsmurfmy 1.4 is rebooting now
23:00.45DaeJeon-Newbiei don't want to run at boot
23:00.55DaeJeon-Newbiehow can I STOP IT?
23:01.06hbsmurfwhat dist?
23:01.12hbsmurfpoundkey or on something else?
23:02.47*** join/#asterisk John-Z (n=lotek@phrank.aus.us.siteprotect.com)
23:02.48De_Monhbsmurf that would depend on why it is starting on boot
23:03.07*** join/#asterisk alamantia (i=anthonyl@nat/digium/x-4742b1e9a2fa3e54)
23:03.10hbsmurfNo, what I wanted to know is what distribution he's running Asterisk on
23:03.21hbsmurfthat way I can tell him if I know the command to stop it loading on boot
23:03.22DaeJeon-Newbietrixbox
23:03.30hbsmurfchkconfig asterisk --off
23:03.44John-ZQuick Questions guys.. Ive setup everying correctly.. like normal for most of my soft phones here.. but a new extension I've configured.. my changes are not taking.. it seems the display name is staying the same after chaging the config files for the phone and reloading.
23:03.47hbsmurfI know enough to be dangerous
23:03.59hbsmurfhave you tried restarting?
23:04.07John-ZMe?
23:04.35hbsmurfyep
23:05.35brodiemJohn-Z, whats up :)
23:05.36John-ZThe phones were once setup for previous employees.. and their display name seems to stay the same even after changing to config and performing a 'reload'
23:05.36John-Zbrodiem: Hows it going?
23:05.36brodiemJohn-Z you know who I am?
23:05.36hbsmurfyou're changing the sip.conf, right?
23:05.40John-ZI do not. Sorry.
23:05.46brodiemJohn-Z apollo?
23:05.52John-ZOh dude.. WHATS UP!
23:05.55brodiemhahah
23:05.56John-ZSmall world.
23:06.01brodiemyea really
23:06.11John-ZIm over at Hostway.. Data Center.. Sys Admin.
23:06.19John-ZThis trixbox is bugging me.. :(
23:06.26brodiemnice
23:06.30brodiemthey got you doing voip over there? :)
23:06.39John-ZYep.. amoungst many other things.
23:06.43John-ZYou still at Apollo?
23:06.54brodiemyup
23:07.00*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
23:07.36[TK]D-FenderJohn-Z : Then you're in the wrong place...
23:08.20John-ZIm in the wrong place, what do you mean?
23:08.29hbsmurfHe's grouchy
23:08.37John-ZI see.
23:08.56John-Zhbsmurf: you talking about /etc/asterisk/sip.conf ?
23:09.00hbsmurfYep
23:09.39DaeJeon-Newbiei tried to restart
23:09.40hbsmurfsweet, 1.4 yells at me when I do sudo su
23:09.47*** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net)
23:09.56DaeJeon-NewbieNOTICE[3140]: http.c:578 http_server_start: Unable to bind http server to 0.0.0.0:8088: Address already in use
23:10.07DaeJeon-Newbiehow can I FIX IT?
23:10.08[TK]D-Fenderhbsmurf : No, I just like it when people read the channel topic when they come in here.
23:10.29hbsmurf[TK]D-Fender:  I still say you're grouchy
23:10.31John-ZOh..
23:10.40John-ZMaybe I should go to #trixbox.. if someone were in there.
23:10.41hbsmurfYes, it isn't the trixbox channel
23:10.52hbsmurfbut still
23:11.11DaeJeon-NewbieBUT I AM NOT USING TRIX
23:11.11*** join/#asterisk emphyrio (n=stryfe@dsl254-076-201.nyc1.dsl.speakeasy.net)
23:11.11DaeJeon-NewbieI REMOVED
23:11.13hbsmurfnewbie: not you
23:11.25hbsmurfnewbie: What are you starting that is erroring out?
23:11.34[TK]D-FenderDaeJeon-Newbie : try asking in #apache .
23:11.57DaeJeon-Newbieit is not apache
23:12.26[TK]D-FenderDaeJeon-Newbie : Well its clearly not *.  Go ask in that programs support channel.
23:12.31hbsmurfnewbie:  What is it?
23:13.22DaeJeon-Newbiehttp://pastebin.ca/280494
23:13.29DaeJeon-Newbieplease have a look
23:13.55DaeJeon-NewbieI am trying to start asterisk
23:14.12hbsmurfInteresting
23:14.24DaeJeon-Newbiei tried to kill all the process
23:14.32DaeJeon-Newbiekillall -9 asterisk
23:14.34hbsmurfIt looks like the manager socket is in use
23:14.39hbsmurfdo a ps ax and look for asterisk
23:14.41hbsmurfor safe_asterisk
23:14.56*** join/#asterisk seele_ (n=seele@208.35.117.246)
23:15.06DaeJeon-Newbiebut again it is restarting
23:15.15DaeJeon-Newbiei was able it kill
23:15.27hbsmurfkill safe_asterisk if it's running
23:15.37seele_hello, how can I manage multiple PBX clients with the same extensions in the same asterisk .... is this posible?
23:15.44DaeJeon-Newbie<PROTECTED>
23:15.50DaeJeon-Newbieautomatic
23:16.19hbsmurfsounds liek safe_asterisk to me
23:16.23_BOBWEEVERseele_: contexts, they are fairly well documented
23:17.32seele_I can make extensions with the same number
23:17.39hbsmurfkind of
23:17.48hbsmurfyou won't use the same numbers for your endpoints, but in your dial plan you can
23:19.30*** join/#asterisk hyperthread (i=hyperthr@c93425d8.virtua.com.br)
23:20.46*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
23:21.26*** mode/#asterisk [+o mog] by ChanServ
23:22.04hyperthreadhello all....Can I use the result of a linux command (using the "system" dialplan command) as a variable value ?
23:22.49[TK]D-Fenderhyperthread : Not really.  Thats what AGI is for.
23:23.02DaeJeon-Newbiehbsmurf: i was modifing the file , but i coult not save
23:23.06DaeJeon-Newbiehttp://pastebin.ca/280508
23:23.29DaeJeon-Newbiehow can I delete .swp file
23:23.34hbsmurfAre you running trixbox?
23:23.35bkruse[TK]D-Fender: i bet you could.
23:23.45bkruse[TK]D-Fender: but your right, thats what agi's for
23:23.47DaeJeon-Newbieno no
23:23.50DaeJeon-NewbieI am not
23:24.03hbsmurfHmmm
23:24.04bkruserm *.swl
23:24.05DaeJeon-NewbieI removed the pre. conf
23:24.06bkruseswp*
23:24.07hbsmurfAre you in as root?
23:24.11DaeJeon-Newbieyes
23:24.22[TK]D-Fenderbkruse : Please point to some doc referencing this possibility.....
23:24.24bkruseor vi it in then hit r for recover
23:24.31hyperthreadDaeJeon-Newbie, I dont like AGI
23:24.40bkruse[TK]D-Fender: i just did it, set var then have the var be a system call
23:24.41bkrusebrb
23:24.50hbsmurfOMG
23:24.53hbsmurfMy 7920 works!
23:24.54hbsmurfin 1.4!
23:25.00Nuggetyay
23:25.12DaeJeon-Newbiehbsmurf?
23:25.18hbsmurfYes
23:25.18hbsmurf?
23:25.24hbsmurfDelete the .swp files
23:25.32DaeJeon-Newbiehow?
23:25.34hbsmurfin /etc/asterisk
23:25.40hbsmurfrm -rf manager.conf.swp
23:25.41[TK]D-Fenderbkruse : that is the opposite of what was just asked.  he asked if you could set a var BASED on the output of a System call.  Not call System passing a var.
23:25.52*** join/#asterisk Rhiliam (n=gary@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com)
23:26.21russellbno, but implementing system as a dialplan function vs. an app would be cool
23:26.23DaeJeon-Newbiei did
23:26.34hbsmurfif you do an ls -l can you see anything like that?
23:26.35DaeJeon-Newbiebut still it is showing the same msg
23:26.35Rhiliamis there a way to do an if/then state within a dialplan?
23:26.48hbsmurfwhat are you getting that message in?
23:26.49hbsmurfvi?
23:26.53DaeJeon-Newbieyes
23:26.56groogsRhiliam:  GotoIf()
23:26.57DaeJeon-Newbiewhen I open
23:26.59hbsmurfTry nano
23:27.02hbsmurfnano manager.conf
23:27.06DaeJeon-Newbieok
23:27.49*** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net)
23:28.12DaeJeon-Newbieyes
23:28.15DaeJeon-Newbiefixed
23:28.17DaeJeon-Newbieno msg
23:28.18hbsmurfSweeeeet
23:28.24hbsmurfI'm not a big vi fan
23:28.32hbsmurfI like simple
23:28.40hbsmurfmy women, simple
23:29.04hbsmurfmy cars, simple
23:29.04hbsmurfmy voip, simple
23:29.04hbsmurfheh
23:29.04DaeJeon-Newbievery good
23:29.12DaeJeon-Newbieman I have another problem
23:29.23hyperthreadI made a shell that do a select in a database and return 1 to true or 0 to false...I want to do a gotoif if the result of the shell is 0...
23:31.01DaeJeon-Newbiei never used nano
23:31.14DaeJeon-Newbiehow can i save the file?
23:31.21hbsmurfctrl-x
23:31.23hbsmurfthen y
23:31.25hbsmurfdammit
23:31.30hbsmurfI just broke the screen on my snom 320
23:31.35hbsmurfnot that I used it
23:31.40hbsmurfit's just sitting in my basement
23:32.20[TK]D-Fenderhyperthread : Go ready up on AGI.
23:33.55*** join/#asterisk brannfenix- (n=brannfen@ip68-230-133-70.ri.ri.cox.net)
23:35.41*** join/#asterisk Marcus_ (n=Marcus@mendelson.ethon.com)
23:39.11*** part/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)
23:39.31robin__szyou know what I think ... an Eclipse plugin for creating Asterisk dial plans would rock somewhat
23:39.37hyperthread[TK]D-Fender, ok...lets try it.... :-(
23:40.39De_MonI want to use conferences in a support center where each call gets a new conference. How can I dynamicaly create the conference numbers ranging from like 800-900
23:40.56robin__szhmmm ...
23:41.01De_MonI'd have to check if the number is in use before putting a new call into the conference
23:41.15De_Monor making sure the callers is 0?  
23:41.23robin__szand typoically, these calls would have just two people per conference room?
23:41.30*** join/#asterisk sloth (n=sloth@cpe-69-203-212-182.nyc.res.rr.com)
23:41.40robin__szand extra people could be called in if required?
23:41.42De_Monyeah, typically
23:41.56De_Monexactly
23:42.09robin__szI dont think you want to do that :)
23:42.38robin__szfrom what I remember, conferencing can load the box
23:42.43robin__szquite heavily
23:42.58De_Moni dont expect more than 5 conferences at a time, and the box has plenty of horsepower
23:43.11robin__szah, 5 would be OK
23:43.43*** join/#asterisk d42 (n=don@124.189.33.60)
23:43.45hbsmurfWell, no transfer button the 7920 does suck
23:43.52robin__szso you have like 10 support staff?
23:44.14De_MonI cant really think of a better way to have more than 2 people on the call at once without using 3way calling which the remote caller's phones dont support
23:44.54robin__szI suspect having one conference room per support person would be easiest
23:45.20seele_how can I host virtual PBXes
23:45.47De_MonI want to use it like parking where it parks in a conference and tells me what # they are in
23:46.01*** join/#asterisk orcimrepus (n=orcimrep@74-130-60-85.dhcp.insightbb.com)
23:46.09hyperthreadthe return of a AGI script is the same of the value of the #? ?
23:46.35hyperthreadsorry, $?
23:46.43hyperthread0 to ok and 1 to error ?
23:50.02robin__szI would have thought playing htem a message, parking them in orbit and then dropping them into a room as soon as an agent presses "next sucker from the queue" would work nicer
23:51.38*** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org)
23:55.31*** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net)
23:55.31*** part/#asterisk dasenjo (n=dasenjo@190.24.176.58)

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.