irclog2html for #asterisk on 20050921

00:00.02Saaibjpablo: que usaste como appliance ? yo uso soekris
00:00.09ariel_jpablo, nadamas 100 persona's.
00:01.18jpabloSaaib: mini atxs, ya que le metiamos tarjetas tdm digium
00:02.09*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
00:02.18jpabloariel_: bueno, eran como 120 extensiones en la oficina principal, con enlaces a 6 sucursales con mas o menos 20 extensiones cada una :-P
00:02.39fiber0ptiLegacy phone systems have the ability to show which lines are in use and who is on the phone. is there a way to do this with asterisk and softphones, polycom and cisco ip phones?
00:02.56Qwellfiber0pti: Those are generally key systems
00:03.12Qwellafaik anyhow
00:03.28*** join/#asterisk SwK (i=vthkif@12-219-144-126.client.mchsi.com)
00:03.38QwellI guess you technically could do it...it'd be a hack though...
00:03.39ariel_fiber0pti, yes, FOP, hint on the polycom's but it should only be for reference. better train people on it's proper use.
00:04.06ariel_fiber0pti, and not all keyed system did that.
00:04.17Qwellprobably alot easier to do with sccp...but, I don't know
00:05.09ariel_argh Cisco's
00:05.30QwellI should play with sccp sometime, just for fun
00:05.52fiber0ptiis that normal for an IP phone to be able to see which lines are in use?
00:06.16ariel_jpablo, mul bien
00:07.06jpabloard: de donde eres?
00:07.09supaigtrAnyone looked at automon using vm email delivery?
00:07.11ariel_fiber0pti, no
00:07.37ariel_jpablo, miami (greengo here)
00:08.03jpabloariel_: oh
00:08.58ariel_jpablo, tran tando de escribire en espanol no es mul fasi para mi.
00:09.41groogsfiber0pti: thats really only used by small systems.. think of having 100 lines.. do you give everyone a huge operator panel that takes up half their desk? :)
00:10.06fiber0ptihaha
00:10.08fiber0ptigood point
00:10.25groogsfiber0pti: its a bit annoying at first, if you're used to a key system
00:10.50fiber0ptinod.. understood..
00:10.53supaigtrfiber0pti: We have 400+ DSS console all over the place.
00:11.13groogsfiber0pti: but wtih *, you can add Voip trunks, which give you lots of room. i use voip for outgoing LD, and local calls if our analog lines are full
00:11.30groogsfiber0pti: the only time people hear a busy signal when trying to dial out is if our internet connection goes down
00:11.38groogs(and all the analog lines are in use)
00:11.52fiber0ptigroogs: thats a good idea..
00:12.04fiber0ptiaight.. time to run. .thanks for your help!
00:12.07groogsi actually do the same for inbound..
00:12.09groogsok then
00:13.03groogsmy last analog line is set to hunt to a voip DID number.. i don't RELY on it (since you should never RELY on voip with a non-dedicated non-private connection), but if all the analog lines are full, and someone calls, it comes in using voip
00:13.37supaigtrAnyone here know if the 411p is 16ms or 64ms of echo can?
00:14.24jpabloariel_: ya veo. tienes un nombre hispano sin embargo ..
00:14.30ariel_16ms
00:14.46supaigtr:(
00:14.57supaigtrI thought the web said 64ms
00:15.00ariel_jpablo, si my padres son parte the cuba
00:15.39ariel_supaigtr, there was a talk about that on the -dev about 3 weeks ago
00:16.44supaigtrariel_: Its caused me major issues with our system.  We went from OK to worse.
00:16.55supaigtrWe tried the vpmspans but that broke DTMF.
00:17.28ariel_supaigtr, humm guess it's time to call digium on there free support or remove the dauther board and test it without.
00:18.38supaigtrI've called and called.  Be promised and promised but its been like a month or so.  We got the 411p in bulk when they first came out.
00:19.04ariel_supaigtr, wow
00:19.14hardwiremother of god
00:19.44ariel_supaigtr, ask if there is any digium employee here and ask them to help.
00:20.40hardwiremy zap channel is hanging up live calls
00:20.44hardwirebecause a busy tone was detected
00:20.48hardwirewtF?!
00:20.56*** join/#asterisk valence (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
00:20.59Qwellhardwire: got callprogress=yes?
00:21.07hardwireof .. course ..
00:21.08supaigtrariel_: There isn't a fix yet.  Basically what I have been told is they are working on software to allow zaptel to use pick spans to e can which may give more tail ms.  But right now it breaks other stuff like DTMF.
00:21.10Qwellln -s callprogress randomlyhangupcalls
00:21.18hardwireactua
00:21.19hardwirelly
00:21.21hardwireits set to no
00:21.36hardwirebusydetect is set to yes :)
00:21.46ariel_supaigtr, have you tried the kb1 echo routin?
00:21.48*** join/#asterisk santiago (n=santiago@63.245.86.145)
00:22.12supaigtrhardwire: I have similar problem when user calls IVR that sets PRI to ring until live person answers.  My PRI trace says * hangs up.
00:22.26hardwiresupaigtr: doh
00:22.27hardwirewell
00:22.31hardwireI have sip to sip hangups too
00:22.31supaigtrariel_: It works great but I'm not sure it'll handle our volume.
00:22.32hardwirehoweevr..
00:22.35hardwireI care less about that
00:22.36ariel_callprogress=no works
00:23.01hardwirerawr?
00:23.07hardwirehmm.
00:23.57hardwirehow many people have problems in multi-lan pbx setups with canreinvite?
00:24.32ariel_hardwire, I don't use canreinvite=yes at all.
00:24.47hardwirejust wanted to take asterisk out of the loop
00:24.52hardwirefor most sip to sip traffic
00:25.10hardwireQwell: I have a pri.. so it should be giving me most of the indications eh?
00:25.19Qwellgot me :p
00:25.26hardwireyou think you are all smart
00:25.28hardwireand then NO!
00:25.35QwellI've just heard that callprogress does what you described sometimes
00:25.39ariel_hardwire, thanks
00:25.55supaigtrhardwire: What is the pri status of the channel when disconnect?
00:26.05hardwiregive me an example
00:26.09hardwireariel_: ?
00:26.22hardwirehttp://pastebin.ca/23482
00:26.26hardwirethats a paste of the hangup
00:26.58supaigtrIn my case its in RING state.  I guess sirus and dell are trying to save money and rather than PRI connect they indicate ring until connect actually happens.
00:27.12hardwirehm
00:27.29hardwireand you get dropped calls due to that?
00:27.30tzangerre
00:27.33hardwirebar
00:27.46supaigtrYea.  iTs weird.
00:28.03supaigtrIs there a busy tone on the line when it hangs up or * just says that?
00:32.06hardwirewell
00:32.11hardwirewho am I gonna call to get a test busy tone
00:32.11hardwireheh
00:32.32L|NUXhttp://pastebin.ca/23486
00:32.46L|NUXcan some one tell me why i don't get fwd iax call on my local extension 1
00:33.00L|NUXwhen ever i dial it say personal extension 1002 is unavailable :(
00:33.24hardwireGot a FRAME_CONTROL (5) frame on channel Zap/2-1
00:33.27hardwirewhat the hell does that even mean?
00:35.54Darwin35linux it means that your phone is not registering
00:35.59hardwireit only happens on Zap channels
00:36.04L|NUXwell i am registered :(
00:36.23Darwin35can you dial outside lines
00:36.59L|NUXDarwin35 : i don't have provider :)
00:37.17L|NUXgot it working :)
00:37.26Darwin35?
00:37.35Darwin35what was the issue
00:37.37L|NUXDarwin35 : i just registered it on another server as well :)
00:37.44Darwin35ok
00:37.50hardwireour new receptionist is a little too cute for words
00:37.52hardwiretherefore
00:37.54L|NUXDarwin35 : how can i regtser a sip on another port in my sip.conf
00:37.55hardwireI shall say nothing
00:38.10L|NUXi did tried register => user:sec@host:port/ext
00:38.16L|NUXbut it won't tregistered :(
00:38.18DarthClueSep 20 19:36:51 NOTICE[3692]: channel.c:2141 __ast_request_and_dial: Don't know what to do with control frame 15
00:38.18DarthClueDon't know what to do with control frame 15
00:38.55Darwin35yes it has
00:39.08Darwin35and the rest of us doing pot
00:39.09L|NUXDarwin35 : hmm
00:39.15Darwin35hold on
00:39.22Darwin35grrrrr
00:39.35DarthCluewhat?  and nobody passed it to me?  that just ain't right!
00:39.41Darwin35just woke from a nap the brain is catching up
00:40.03Darwin35Darth sorry snooz you loose
00:40.05QwellL|NUX: what did your provider say when you called them?
00:41.05*** join/#asterisk dalabera (n=dalabera@adsl-146-34-251.mia.bellsouth.net)
00:41.47dalaberadoes anyone have a sip/iax asterisk box in canada?
00:42.05Darwin35why in canada
00:42.09L|NUXQwell : PTCL blocked port 5??? series :(
00:42.10Darwin35the US to good for oyu
00:42.28dalaberaI know...
00:42.56Sedoroxyes.. actually
00:43.02L|NUXDarwin35 : hmm
00:43.10L|NUXDarwin35 : send me visa :P
00:43.15Sedoroxgot another one here they will be colo'd to replace the exsisting one actually
00:43.29Darwin35?
00:43.44Sedorox[20:41] dalabera does anyone have a sip/iax asterisk box in canada?
00:43.58*** join/#asterisk santiago (n=santiago@63.245.86.145)
00:44.58Darwin35well fine
00:45.01Darwin35btw
00:45.31L|NUXhmm
00:45.32L|NUX:)
00:45.50L|NUXDarwin35 : what to do to connect on differnet port then 5060 :)
00:46.29dalaberasedorox -> anything to say?
00:46.31hardwirewell
00:46.34hardwireI think I found the issue
00:46.38hardwirecallprogress and busydetect
00:46.51hardwirewere callsing FRAME_CONTROL + subclass AST_CONTROL_ANSWER and AST_CONTROL_BUSY
00:47.40Sedoroxabout....?
00:48.04hardwiresex
00:48.05hardwirebaby
00:48.09dalabera* in CA
00:48.10hardwirelets takl about you
00:48.11Darwin35Linux I am looking just wait
00:48.11hardwireand me
00:48.18hardwirelets talk about all the good things.. and the bad things..
00:48.50L|NUXok
00:49.17Sedoroxumm
00:49.26Sedoroxasterisk in canada is no different then it is in the US
00:49.34Sedoroxor in Budapest for that matter...
00:49.40dalaberaLOL
00:49.47hardwireheh
00:49.47dalaberaI know that...
00:49.50Sedorox:p
00:50.00Sedoroxor you mean latency?
00:50.07hardwireits slightly different :)
00:50.16dalaberathe thing is I want to test a couple of numbers I have...
00:50.19DarthClueum, well actually, in canada, asterisk only works if you talk dirty to it
00:50.28hardwireused under a different locale or TZ :)
00:50.37Sedoroxlol DarthClue
00:50.45hardwirecanadians don't talk dirty.
00:50.55Sedoroxdalabera, ok....?
00:50.58*** join/#asterisk J[SS] (i=ph33r@smartserv/ceo/chaoscon)
00:50.59hardwireI have seen them try.. they fail.
00:51.21DarthCluein budapest, you have to give it tea, in russia, vodka, in oklahoma, you have to use shotguns to make it work
00:51.40hardwirein mother russia.. asterisk calls you..
00:51.46*** join/#asterisk Tili (i=Tili@203.101.168.27)
00:51.51dalaberathat's ok...
00:52.10Sedoroxlol
00:52.20Sedoroxactually.. anywhere asterisk can call you :p
00:52.22Sedoroxbut anyway
00:52.26Sedoroxdalabera, what do you need?
00:53.21dalaberaan asterisk box I can hook up to call a friend at 647 area code!! :-)
00:53.32litagesethk: when setting up an ssh tunnel, the format is this, right?:    ssh -N -L <port on localmachine to forward from>:<ip addr on remotemachine to listen on>:<port on remotemachine to listen on> user@remotemachine
00:53.37litages/sethk//
00:53.40Sedoroxoooooo you want us to provide you a box to call through?
00:53.49Qwelllitage: looks right
00:53.58dalaberaLOL
00:54.20Sedoroxor someone to host a box for you in CND?
00:54.28dalaberanevermind...
00:54.44Sedoroxmm ok... sorry... just not understanding clearly
00:54.44Qwellyour "remotemachine to listen on" is quite wrong though.  "ip:port of machine to send connection to" is more accurate
00:54.48*** join/#asterisk imcdona (n=aspersio@c-24-19-90-119.hsd1.wa.comcast.net)
00:55.31L|NUXQwell : i created context and then try to register it through register => @BrainTel/1
00:55.36L|NUXbut its still not working
00:55.47Qwellthat shouldn't work
00:55.48*** part/#asterisk imcdona (n=aspersio@c-24-19-90-119.hsd1.wa.comcast.net)
00:56.05L|NUXumm
00:56.09L|NUXthen what to do :D
00:56.17*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
00:56.28QwellYou need to find out if they're sending any data back, and if so, on what port
00:56.38Qwellif its not coming back on 5060, its probably not going to work
00:56.48L|NUXhmm
00:56.54QwellDidn't I say that like two days ago?
00:56.59L|NUXhow can i check this by sip debug peer :)
00:57.05QwellYou don't
00:57.17Qwellif your firewall is blocking it, and/or asterisk isn't listening on that port, it'll never see it
00:57.28L|NUXhmm
00:57.32litageQwell: i'm trying to forward packets from local:6000 to remote:6001, with ssh running on remote:9000. however, i see no activity with this cmd:   ssh -N -L 6000:127.0.0.1:6001 user@remote -p 9000
00:58.11Qwellit'll ask for password, and thats it
00:58.19L|NUXtcpdump: verbose output suppressed, use -v or -vv for full protocol decode
00:58.19L|NUXlistening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
00:58.20L|NUX10:42:26.960132 IP 202.5.145.14.5060 > 203.128.7.14.8891: UDP, length 376
00:58.20L|NUX10:42:27.960878 IP 202.5.145.14.5060 > 203.128.7.14.8891: UDP, length 376
00:58.20Qwellthen it should background itself
00:58.24L|NUXthis is what i getting
00:58.31Qwellno, thats what you're sending
00:58.48L|NUXahhh
00:58.56Qwellunless your IP is the 203. address
00:59.16L|NUX10:43:21.640981 IP 202.5.145.14.3967 > 203.128.7.14.8891: UDP, length 20
00:59.16L|NUX10:43:22.441467 IP 202.5.145.14.3967 > 203.128.7.14.8891: UDP, length 20
00:59.16L|NUX10:43:24.128221 IP 202.5.145.14.9204 > 203.128.7.14.8891: UDP, length 20
00:59.16L|NUX10:43:24.328692 IP 202.5.145.14.9204 > 203.128.7.14.8891: UDP, length 20
00:59.16L|NUX10:43:24.529128 IP 202.5.145.14.9204 > 203.128.7.14.8891: UDP, length 20
00:59.16Qwellso, like I said...you need to call them, and find out whats happening to the return packets
00:59.19*** part/#asterisk jpablo (n=jpablo@dsl-200-78-66-23.prod-infinitum.com.mx)
00:59.26Qwelldon't paste in here
00:59.32L|NUXthis is now when i get registered though eyebeam
00:59.33Qwelland again, thats YOU sending TO them
00:59.33L|NUXsorry
00:59.42NuggetAnd this, kids, is your brain on Linux.
00:59.43Qwellunless your IP is the 203. address
00:59.49*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
01:00.20QwellIS your IP the 203. address?
01:00.51L|NUXQwell : well provider addy
01:00.55L|NUXmy sip is 202.5
01:01.02QwellI still see no return packets
01:01.12Qwellwhich means eyebeam didn't register, or you didn't show all of the output
01:01.30L|NUXwell its working
01:01.36L|NUXcan i give you user id and passs
01:01.36L|NUX?
01:01.38Qwellno
01:01.43L|NUXaww come on
01:01.48L|NUXtry ti
01:01.49*** join/#asterisk ChulJin (n=chuljin@adsl-68-121-94-235.dsl.irvnca.pacbell.net)
01:01.55L|NUXmay be you will get what i can't see
01:01.56L|NUX:)
01:01.58QwellI'd just steal the account and use all your minutes...
01:01.59Qwellno offense
01:02.08L|NUXwell its won't have minutes :D
01:02.09L|NUXhehe
01:02.15L|NUXQwell : and i trust on you
01:02.18QwellWhat works when you use eyebeam, exactly?
01:02.24Qwelljust outgoing, or does incoming work too?
01:02.26L|NUXwell its working
01:02.30L|NUXincoming :)
01:02.42L|NUXand outgoing on 210XXXXX numbers
01:02.50L|NUXwhere 210 numbers are braintel numbers
01:02.52L|NUX:)
01:02.57Qwellrun a longer tcpdump, from asterisk and eyebeam, wait for the incoming packets to show up
01:02.58L|NUXand they are DID :D
01:03.00Qwell~pb
01:03.01jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
01:03.09L|NUXok
01:03.39L|NUXjust going no incoming packets :(
01:03.47Qwellwith asterisk or eyebeam?
01:04.18L|NUXeven when i dial from my ptcl number
01:04.29L|NUXcan i pb my eyebeam log ?
01:04.36*** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
01:04.36Qwellthe tcpdump log
01:04.59L|NUXtcpdump have all going not incoming :(
01:05.27L|NUXhttp://pastebin.ca/23488
01:05.29L|NUXeyebeam
01:06.21*** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
01:06.22Qwellthis is an asterisk log?
01:06.38L|NUXQwell : its eyebeam log :(
01:06.42L|NUXnot asterisk
01:06.56*** join/#asterisk gclark (n=gclark@cpe-069-132-101-069.carolina.res.rr.com)
01:07.00Qwellok, and where is the tcpdump log of eyebeam registering?
01:07.13L|NUXwait
01:07.19*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:07.22gclarkHello all... I have a question regarding CVS + IAX
01:07.55gclarkI keep on receiving this error message
01:07.56gclarkchan_iax2.c:5586 update_registry: Restricting registration for peer '3031' to 60 seconds (requested 0)
01:07.59gclarkAny ideas?
01:08.01Qwelleyebeam is telling them to take calls on 9204
01:08.07L|NUXhttp://pastebin.ca/23489
01:08.08Qwellerm, send calls to
01:08.22L|NUXhmm
01:08.24Qwellhey look, incoming packets...and I was right
01:08.29miahis there a context setting for sip peers to allow them to dial out?  i keep getting messages from asterisk saying im not allowed to dial out, my  peer type is set to 'friend'
01:08.30L|NUXyou know 202. is also my gw
01:08.31L|NUX:)
01:08.34Qwellnow show me the tcpdump log of asterisk registering
01:08.43L|NUXwell dude
01:08.53L|NUXi don't have access on chat.brain.net.pk
01:08.55L|NUXi have access on my own
01:08.56L|NUX:)
01:09.10L|NUXwhoses logs i already showed you
01:09.11L|NUX:)
01:09.23Qwellthese logs are exactly what I asked for (4 times)
01:09.28Qwellnow show me the same thing, but for asterisk
01:09.36L|NUX:)
01:09.46L|NUXwait
01:11.10L|NUXhttp://pastebin.ca/23490
01:11.25Qwella tcpdump of asterisk registering
01:11.30L|NUXawww
01:11.37QwellI've already seen this a bunch of times
01:12.04ChulJin~dundi
01:12.05jbotit has been said that dundi is http://www.dundi.com
01:12.06L|NUXok
01:14.04ChulJinhas the excitement of dundi largely worn off?
01:14.41DarthCluei'm not sure dundi was ever very exciting for most of us.
01:14.47gclarkRestricting registration for peer '3031' to 60 seconds - IAX Peer Registration - any one seen this?
01:15.44ChulJinfair enough :P
01:16.07L|NUXhttp://pastebin.ca/23491
01:16.09L|NUXQwell : http://pastebin.ca/23491
01:17.06Qwellhey look, 9204 again
01:17.15L|NUXhmm
01:17.16Qwellis eyebeam running?  close it
01:17.19L|NUXyeah
01:17.20L|NUXi did
01:17.28L|NUXclosed
01:18.03L|NUXnow not getting any thing from 203.....
01:18.22Qwellyou should only really get something back, when you're sending something
01:18.32L|NUXyeah
01:18.34Qwellwhy the hell are they trying to send back on 9204?
01:18.34L|NUXno idea
01:18.37L|NUXwhy i am not getting
01:18.43L|NUXno idea :(
01:18.57L|NUXbecause 5??? blocked by PTCL
01:19.19Qwellwhat is ptcl, why is it blocked, and why does that matter?
01:19.26L|NUXand PTCL giving bandwidth to all ISP's in pakistan
01:19.37L|NUXPakistan Telecommunication Company Limited
01:19.43Qwellso...
01:19.50Qwellif 5060 is blocked on your ISP
01:19.55Qwellwhy not have asterisk listen on 9204?
01:20.00L|NUXwell i am direct on fiber :)
01:20.08L|NUXwell i can connect on 5060 :)
01:20.15L|NUXi have flag link :)
01:20.15QwellI don't think chan_sip can listen on multiple ports
01:20.27L|NUX:(
01:20.46Qwellget a new provider
01:20.55L|NUXhmm
01:20.56L|NUXk
01:21.28Nuggetand fix your enter key.  :)
01:25.45Vcoso is VoIP still brutally illegal in Pakistan?
01:26.30*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
01:28.02mog_homewhy?
01:29.36mog_homewell i guess i know why
01:29.39*** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3781706.sympatico.ca)
01:29.43mog_homebut still lame
01:36.14*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
01:36.52UberbotHi all.
01:39.41UberbotI seem to be having difficulty playing a gsm file in the background... the console displays a message saying that it can't open the stream file.... Help?
01:39.56Uther_Pdont put the extension
01:40.12Uther_Pif the file is called  menu.gsm,   just put menu
01:40.16TiliVco: there is some level of reduction in that brutality. I see mobile carriers now using VoIP for international calls.
01:40.36UberbotThanx, I'll try that, Uther_P
01:41.29UberbotSep 20 16:12:08 WARNING[14237]: file.c:475 ast_openstream: File "/var/lib/asterisk/sounds/diehl_intro" does not exist in any format
01:41.48UberbotBut the file does exist....
01:42.14UberbotAny other ideas?
01:42.39*** join/#asterisk ZX81 (i=matt@222-153-100-242.jetstream.xtra.co.nz)
01:42.47file[laptop]OMG IT'S ZX8!
01:42.50file[laptop]ZX81!
01:42.52file[laptop]<PANIC>
01:42.54ZX81:D
01:43.00ZX81hehe
01:43.05ZX81how are you?
01:43.15ZX81wanna be my beta monkey
01:43.20ZX81:D
01:43.22ZX81kewl
01:43.23file[laptop]not particularly, I'm in bed :P
01:43.27ZX81freevoip.gedameurope.com
01:43.32ZX81ah
01:43.35ZX81lol
01:43.37ZX81:)
01:43.39*** join/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net)
01:43.42ZX81kk
01:43.57file[laptop]oh no, what's that?
01:44.07pbdEvening, all.
01:44.10ZX81free realtime
01:44.12ZX81:)
01:44.39pbdPop quiz: Anyone seen any problems when the message log file grows larger than 2G under 1.0.7?
01:45.04file[laptop]crazy
01:45.39pbdYes, I know it is. :)
01:46.11denonpbd: that issue has been around for a long time
01:46.22denonset up a logger to deal with it
01:46.42Uberbotecho > /var/log/asterisk/cdr-csv/Master.csv  :-D
01:46.56pbdI know, I've seen it referenced a few places- I'm just looking for a bug tracker number or something I can show a client, without having to pull apart the code and cross reference glibc.
01:48.24pbdI'd set up a logrotate for it- but they hit it in 10 hours.. they've got debugging output going to a full log, as it's a new install and they're searching for errors- I've no issue with that, but I want to see if anyone has it 'in progress' somewhere.  Technically, a new glibc would probably solve it- but I hate to upgrade for the sake of upgrading.
01:48.52*** join/#asterisk spackle (n=spackle@209.234.83.19)
01:49.25denondunno, the "problem" has been around forever
01:49.47*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
01:49.56pbdThat's the 'problem'- it's been around too long. :)
01:50.06*** join/#asterisk dos000 (n=dos000@i216-58-26-59.cybersurf.com)
01:50.35*** join/#asterisk ZX81 (i=matt@222-153-100-242.jetstream.xtra.co.nz)
01:50.44denonoh I dunno, there comes a point when having a 2GB log file may be the admin's problem, not the software's :)
01:51.18Nuggetheh
01:51.21pbdcouldn't agree more.  One thing that bugs me is that it's referenced under ext2 filesystems- in this case, we're running on ext3.
01:52.26pbd*technicallly* (and tested) the fs can handle it.. it's the libraries referenced by Asterisk that can't deal.
01:53.12*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
01:53.50mog_home2gb log files def sounds like you need to rotate ocasionally
01:54.03pbdIn this case, hourly, mog.
01:54.05Vcoor at leaset *once*
01:54.21pbdAlready set up for daily.  Hit 2G in ten hours.
01:54.28mog_homeyou fill 2 gigs in hours?
01:54.34mog_homehow many calls
01:54.36pbdFun, isn't it?
01:54.39mog_homeand what are youuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuu doing?
01:54.52mog_homethats a bad vnc
01:54.54Vcohow many calls or what is broken and spitting out that many messages?
01:55.07pbdNot that many- the issue is, the client wants to see full tracing- full debugs on, all the bells 'n whistles.
01:55.09mog_homeyeah
01:55.25pbdSo far, he's not doing pri debugging, but the rest...
01:55.26mog_homestill shouldnt be 2 gigs
01:55.31redder86can't logrotate be run hourly?
01:56.00mog_homeunless it is a rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrreally busy/bugggggggggggggggggggggggggggggggggggggggy switch
01:56.02pbdCallcenter environment.  30 agents.  20 active calls.  Iax2 debug, verbose >=3, debug >=3 ... you get there nice and quick.
01:56.27pbdI'd almost like to use it for a random number generator. Or an encryption dictionary. :)
01:57.23pbdLogs are clean- I'm now recommending we *NOT* log debug messages.  But I'd love to see an article to reference that says 'You just can't write out a 2G log file and expect it to work'.
01:57.55pbdLogrotate can be run every minute if necessary- but it makes for no fun debugging with logs
01:58.24mog_homewell 2gig limit is fairly universal
01:59.37pbdExcuses, excuses.  I've seen articles claiming it's been fixed in ulimt.. in ext3.. in newer libc's.. nothing saying 'In Asterisk, it will blow cores' (which it does).
01:59.39redder86are they debugging to hunt down a specific problem?
02:00.10pbdNo, not yet- just the first turnup week of a new callcenter.. It pays to have the logs, since the agents are still getting used to it.
02:00.41pbdI'd expect we can turn it off within a week at most- but we're going to have to turn it off earlier, obviously.
02:00.50redder86that's a lot of logging to go through, though, any time you need to do it
02:02.01pbdAgreed.  If the servers were less powerful, it would probably even trigger performance issues- but they're not using LVM, and they are twin Xeon rocket ships.
02:02.53Tiliwhat is the max network lag VoIP can take with iLBC
02:03.10*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
02:03.36pbdIn this case, Asterisk even has code to auto rotate on SIGXFSZ.. but it's a busy server, and it couldn't complete the rotate before it tried to write, (different thread, I think), and blew core.
02:03.39L|NUXshido6 : hey
02:03.45L|NUXshido6 : quick question
02:03.59L|NUXshido6 : can we bind * on multiple ports ?
02:06.00Nuggetasterisk can bind multiple ports, but a lot of people seem to have problems with it when used that way.
02:06.14Nuggetat one time, perhaps still, it was flaky if you had aliased interfaces
02:06.18*** join/#asterisk tengulre (n=tengulre@61.185.238.166)
02:06.41Nuggetoh, you mean ports.  I was thinking interfaces.
02:06.43mog_homei mean log files
02:06.52mog_homewe run busy ship at digium
02:07.01mog_homeand our master.csv is not that big
02:07.09mog_homeand has been running for a while
02:07.11denonhe said he was in full debug
02:07.20mog_homeeven so
02:07.22L|NUX:)
02:07.26mog_homeit is quite a feat
02:07.30pbdmog: it's not the core that's that big.. it's the full log file itself.  Theoretically, asterisk should have trapped on SIGXFSZ, and rotated the log.. but the core is all over the place with failures in logger_rotate.
02:07.31denonfull debug on calls for a couple hundred people wouldnt take long
02:07.37mog_home2 Gigs of text is ginaromus
02:07.45mog_homeright i understand
02:07.51mog_homei meant log file
02:08.01Nuggetdunno, we generate at least a half gig of text in here every time someone mentions mysql.  :)
02:08.09denonhaha
02:08.17file[laptop]my IRC client crashes every few days because it can't handle #asterisk
02:08.18mog_homelol
02:08.20denonmysql sucks, db2 rocks
02:08.28pbdMy guess is that the threads walked over themselves during the logger rotate on SIGXFSZ.. it was moving pretty quick.
02:08.31mog_homethats why you have to flush irc client file
02:08.39file[laptop]but I'm lazy
02:08.48likwid--file, -rw-------    1 likwid   likwid   131535037 Sep 20 20:51 .naimlog/Freenode/#asterisk.html
02:08.49mog_homeeww
02:08.51mog_homethats nasty
02:08.52pbdWho's up for programming a Microsoft Access backend for Realtime?
02:08.53spacklethere is a new entry into the DBMS holy wars
02:09.01file[laptop]pbd: that's hot
02:09.03mog_homeif the money was right pbd
02:09.14mog_homeid say half a mil
02:09.16Sedoroxand we're talking a lot of money...
02:09.30mog_homeand only because we would have people calling us to support it
02:09.34denon33,213,470 #asterisk.freenode.log
02:09.44denonlooks like I've rotated it "recently"
02:10.00pbdThe Asterisk Summer of Code that Shouldn't Be!
02:10.24mog_homelol
02:10.38Vcofuck...unholy is one word for it....
02:10.38Nuggetheh
02:10.58denong729?
02:11.05Hmm-homei wanna love somebody, love somebody like you
02:11.09e-HernickAsterisk: now driven by the most powerful redmond-industry standard embedded database since the windows registry: MS Access XP.
02:11.26Hmm-homepdb thats not a problem
02:11.43Nuggetmicrosoft access sucks because it fails to do what it's supposed to do.  mysql sucks because it does exactly what it was designed to do.  it's a subtle but significant difference.  :)
02:11.52mog_home209 715.2 pages of text roughly = 2 gig log file
02:11.56pbdbrb
02:12.03Sedoroxahahah Nugget
02:12.51e-HernickI disagree. MySQL rules. When you don't need a fully-featured database, MySQL is great.
02:13.03Nuggetexcept it isn't.  :)
02:13.27Nuggetmysql lacks a lot more than features.  it lacks sanity, safety, and design.
02:13.42e-HernickHowever, it works great for many applications.
02:13.49NuggetI couldn't disagree more
02:13.53e-HernickIt scales much better than other open-source competitors.
02:14.00spackleVIM
02:14.04Nuggetthat's not true either
02:14.34e-HernickDo you believe that say, Wikipedia, would be better served by another OSS DB ?
02:14.40Nuggetabsolutely
02:14.46e-HernickWhat would you propose, then ?
02:14.49e-HernickPostgres ? Firebird ?
02:14.54Nuggetpostgresql, if they want to stay with something costless
02:15.03e-HernickBut Postgres is much slower than MySQL.
02:15.07Nuggetno it isn't.
02:15.10e-HernickThey would need much more hardware.
02:15.19Nuggetespecially for large tasks which need all those "full features" you dismiss
02:15.25spacklee-Hernick, do you really believe that?
02:15.56e-HernickIt has been my experience that for most web applications, which require high read performance and less write performance, MySQL performs better than Postgres.
02:16.04Nuggetmysql is only faster for the most rudimentary and irrelevant tasks.  for anything large or complicated its limitations ruin you far more than its "select *" speed helps
02:16.10e-HernickCertainly, for many other types of applications, Postgres is faster.
02:16.29e-HernickHowever, those tasks you call "rudimentary" and "irrelevant" are enough to build large-scale web applications.
02:16.36Nuggetand, of course, there's the persistant problem of mysql doing things totally wrong and in unsafe ways
02:16.43e-HernickCertainly, the philosophical value of the code is tainted.
02:16.48NuggetSure, you can mow the lawn with scissors too.
02:16.59e-HernickConsider the exact case of Wikipedia.
02:17.04e-HernickWould you advocate a switch to Postgres ?
02:17.13pbdWhere's Brian when you need him.  NEXT.
02:17.21spackleheh
02:17.33Nivexugh... I've been dealing with the headache of adminning a MySQL setup lately
02:17.43e-HernickThe toy database MySQL is more powerful than you give it credit for.
02:18.25NuggetI would go so far as to say that in just about any case, postgresql is more suitable than mysql.   That doesn't always translate into sufficint motivation to migrate, though, which depends on human and time constraints which are independent of the technical merits of the database.
02:18.48e-HernickWhat is holding Postgres back, then ?
02:18.56e-HernickWhy is MySQL so much more popular and widespread ?
02:19.06NuggetMy complaint is not that it's a toy, but more that mysql is recklessly designed and that leads to terrible coding habits and risk of silent data corruption.
02:19.28NuggetWhy is Windows so much more popular and widespread than other alternatives?
02:19.32rozoMySQL could run on windows long before Postgres could
02:19.38rozofree SQL server for Windows
02:20.09NivexI've half jokingly said to my boss that we should move our app to Postgres, but he's hung up on the fact that MySQL is faster.
02:20.13spacklewhy is VHS so much more widespread than betamax?
02:20.16NuggetOne of the things holding postgresql back is the legions of ignorant mysql users who have never used anything else and have no idea how crappy mysql is by comparison to the alternatives.
02:20.39Nuggetthey're loud and they write a log of unfounded gushing statements like "mysql is so much faster than postgresql"
02:20.43Nuggetand "mysql is so much simpler"
02:21.27e-HernickWell, I myself use both systems. I can see the advantages of Postgres, and I use it accordingly, when it is called for.
02:21.40NivexWe're finally migrating our data from MyISAM to InnoDB in hopes that it will help.
02:21.44e-HernickHowever, I still believe that MySQL is faster for web-style applications that use a database for simple queries.
02:22.01e-HernickNow, that's based on benchmarks I have seen over the years.
02:22.07Nuggeta flat file is faster still.
02:22.18e-HernickPerhaps every single benchmark out there is biased and paid for the evil MySQL conspiracy.
02:22.46e-HernickCertainly, but by the time you give a flat-file filesystem DB the kind of functionality and structure you get for free with the most rudimentary SQL database..
02:22.56e-HernickPerformance isn't everything.
02:23.01*** join/#asterisk salviadud (n=dude@201.135.3.68)
02:23.08salviadudhey, i want to test my FWD
02:23.09spacklepersonally I'm just glad there are choices to argue about.  It wasn't long ago when there were slim pickings for any DB
02:23.14Nuggethey, now you're making my argument for me.
02:23.23Nivexe-Hernick: agreed, but it's hard to get PHB's on board with anything else.
02:23.30salviadudcould someone send me a call please? 651692
02:23.59Nuggetin reality, mysql's oft-touted speed benefit completely dissapears as soon as you need something more useful than "select * from".
02:24.10NuggetIt's so easy to outgrow, it seems insane to settle for it at the beginning
02:24.15e-HernickBut many applications are just that.
02:24.28salviadudTHANX!
02:24.31Nuggetand they'd be better off using postgresql.
02:24.41*** join/#asterisk jcohen (n=jc@66-215-19-31.dhcp.gldl.ca.charter.com)
02:24.48Vcofuck that......pencils and graph paper...
02:24.53Nuggetbecause as soon as they do even one insert, they've outgrown mysql's rudimentary and ill-conceived data handling limitations
02:25.14e-HernickSo you say that the simplest app should be built with Postgres ?
02:25.29e-HernickThe second you make that choice, you severely limit your choice of application hosting providers.
02:25.53e-HernickFor a simple application, deviation from the standard means high hosting costs. Diversity will not be tolerated. Conform. Conform.
02:25.55Nuggetwhatever synthetic benefit mysql offers on table-locked selects are completely overshadowed the first time your DBA or an application error accidently tries to insert 50 characters into a char(40) or "february 31st" into a date field.
02:25.57jcohenhi I'm new to asterisk, I am getting this error, Rejected connect attempt from 66.234.228.170, request '5555555555@voicepulse-in' does not exist, what does it mean?
02:26.22DarthCluethe extension 5555555555 in context voicepulse-in doesn't exist.
02:26.26pbd"if you choose not to decide, you still have made a choice"
02:26.55e-HernickThere remains one question.
02:27.02e-HernickHas there recently been a benchmark of Postgres VS MySQL ?
02:27.04DarthCluee-Hernick, while that is true that most providers don't offer anything but mysql or mssql you can still choose to go with something else
02:27.07pbdjcohen: Something or someone is dialing 55555555, and it's hitting your voicepulse-in context.. and it doesn't match.
02:27.15jcohenDarthClue, so that would mean my label in extensions.conf should be [5555555555@voicepulse-in] ?
02:27.54*** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
02:27.55Qwellany good/bad experiences with an spa-3000 and *?
02:27.55e-HernickI am using a pair of SPA-3000 with *.
02:27.55pbdQwell: Good experiences here.
02:27.55*** join/#asterisk Samoied (n=Samoied@200.101.233.182)
02:27.55salviadudi own a spa 3000
02:28.03pbdIt's one handy box.
02:28.04e-HernickThey work great. I'm interfacing one to a Norstar CICS.
02:28.05NuggetI wish I'd gotten an spa-3000 instead of this tdm400p, that's for sure.
02:28.05salviadudi haven't bridged asterisk yet...
02:28.14salviadudbut, thats cause i haven't been trying
02:28.19salviadudit still works though.
02:28.23mog_homeaww nugget whats wrong with the tdm400p?
02:28.26e-HernickYes, extermal self-contained ATAs are a good thing.
02:28.33DarthCluejcohen, no.  the context should be [voicepulse-in] and then you need an extension exten => _X.,1,DoSomethingHere and an s extension to catch everything that comes into that context
02:28.44Nuggetit just stops answering the phone every few weeks and I have to powercycle the box to fix it
02:28.57mog_homethat was a bug fixed months ago
02:29.05jcohenDarthClue, currently my extensions.conf file is:
02:29.08mog_homemark introduced it by accident and fixed it
02:29.09Nuggetin firmware or in zaptel?  I'm running cvs head from a month ago or so
02:29.12mog_homezaptel
02:29.14e-HernickPCI FXO and FXS ports are not such a good idea.
02:29.15DarthCluejcohen, use pastebin
02:29.17Nuggetit's been happening since I bought the card a year ago
02:29.19DarthClue~pastebin
02:29.20jbotsomebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca/
02:29.23jcohenok
02:29.23mog_homethere was a counter that has not been reset
02:29.34mog_homeyouve had the card for over a year
02:29.36Nuggethave you also fixed the bug where I can't reboot the machine?
02:29.55jcohenDarthClue, http://pastebin.com/369616
02:30.38jcohenits a simple extensions.conf file with 4 lines
02:30.58DarthCluejcohen, change the extern to exten on every line, add exten => _X.,1,GoTo(s,1) and see what happens
02:31.10jcoheneek, spelling error
02:31.29e-Hernickdown with extensions.conf, up with AEL
02:31.35mog_homeextern....
02:31.52DarthCluedid they get the bugs in ael fixed yet or is it still crap?
02:32.18e-HernickWell, I've got a small-scale production system running 1.2 beta 1 and AEL
02:32.23jcohenhmm that didn't fix it
02:32.37mog_homewhats wrong with it?
02:32.39DarthCluejcohen, did you add the line i told you to add?
02:32.39e-HernickIt works well for me, no crashes, no problems with the 450 line AEL program I wrote for it
02:32.46mog_homeits very picky on syntax
02:32.49e-Hernickoh yeah
02:32.51mog_homebut i wasnt aware of bugs
02:32.52e-Hernickit's very very picky
02:32.58DarthCluee-Hernick, well, that's better than it was in the beginning
02:33.03jcohenI added it to the end
02:33.05jcohenof the file
02:33.13e-Hernickand it doesn't tell you what's wrong with the syntax
02:33.19e-HernickI figure the parser will get better
02:33.28ZX81oh
02:33.29ZX81hello
02:33.30ZX81:)
02:33.34ZX81I'm still here
02:33.36ZX81:D
02:33.48DarthCluejcohen, it needs to be added to the [voicepulse-in] context
02:33.49e-HernickWhat I think needs to be done though is to remove the AEL parser from asterisk, and instead write it in a higher-level language that produces an intermediary file to be parsed by asterisk
02:33.57EquinoxWhat is AEL?
02:34.04ZX81Asterisk Extension Logic
02:34.07e-HernickIt's the Awful Extension Language.
02:34.08twistedwheee
02:34.10mog_homeheh
02:34.11EquinoxLaugh
02:34.13ZX81like a programming language
02:34.18EquinoxIs it the same stuff in extensiosn.conf on 1.0.x?
02:34.19Sedoroxpbx ~ # uname -a
02:34.19SedoroxLinux pbx 2.6.13-ck5s #2 SMP Tue Sep 20 14:00:39 EDT 2005 i686 Pentium III (Coppermine) GenuineIntel GNU/Linux
02:34.23Sedorox:)
02:34.24ZX81Equinox: nah
02:34.28mog_homebkw_ and friends wrote some things like that i thought
02:34.35jcohenok here is the new file
02:34.36jcohenhttp://pastebin.com/369618
02:34.38mog_homeres_perl, res_javascript etc
02:34.47e-HernickIt replaces extensions.conf; it's way better, but it looks like the sterile bastard child of extensions.conf and perl
02:34.49mog_homeits a comma
02:34.51mog_homeyou need a dot
02:35.03e-HernickAnd it has all the syntax-pickiness of python
02:35.09e-Hernickwithout the readability
02:35.14mog_homeor i need to go to the doctor for an eye exam
02:35.24e-HernickBut it rules, compared to extensions.conf
02:35.34e-HernickI long for AEL 2.0
02:35.36fileZX81 is fun to talk to on the phone
02:35.40mog_homejust too much typing
02:35.46DarthCluejcohen, the line should be exten => _X.,1,GoTo(s,1)
02:35.52*** join/#asterisk litage (n=nick@203.201.99.52)
02:35.58ZX81:D
02:35.59mog_homei need to script vim to add exten => to every line for me
02:36.02jcohenoh a . not a ,
02:36.09mog_homeexactly
02:36.30e-HernickSo, anybody got a SIP conference call system ready to use ?
02:37.06jcohenawesome that did it
02:37.30DarthCluejcohen, just send me a paypal donation
02:37.31mog_homerock on
02:37.35mog_homelol
02:37.48DarthCluei'll give mog his share when he comes to cluecon
02:37.53mog_homelol
02:37.57mog_homei wanted to go
02:38.01mog_homei just had to see folks
02:38.24salviadudcluecon?
02:38.31mog_home~cluecon
02:38.33jbotit has been said that cluecon is http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II.
02:38.33DarthCluewell, i just had to go and see all the cluecon freaks...er, i mean geeks
02:38.45JunK-Ymog_home: :1,$s/^/exten =>/g?
02:38.50salviadudohhh
02:38.50JunK-Y~cluecon2005
02:38.52jbotextra, extra, read all about it, cluecon2005 is http://www.midsouthmarketplace.com/~krice/gallery/
02:38.58mog_homebingo , but on the fly junky
02:39.05salviadudis it the same as astricon?
02:39.12mog_homeno
02:39.15mog_homeit was this summer
02:39.25jlewisis astricon worth going to?
02:39.31e-HernickOkay, here comes an asterisk-related question. I am using call forwarding to track my mobile users with cell phones. They forward their own cell phones to my main number. The IVR answers, and sometimes will transfer the call out to the cell phone, and the call can get stuck in an infinite loop. Right now, I'm using callerID to detect calls that are looping, and process them accordingly.
02:39.32mog_homeyes
02:39.39JunK-Ymog_home??
02:39.46mog_homeit tons of fun /informative
02:39.46e-HernickHowever, is there a way that I can tell the calling-out context that it is looping ?
02:40.07jcohenok so now I'm getting a file doesn't exist error for the playback file that is in /var/lib/asterisk/sounds/
02:40.09jcohenhmm
02:40.22DarthCluejcohen, take out the .gsm you have in there
02:40.23mog_homedoes it exist in right format?
02:40.28e-Hernick.gsm is evil
02:40.31e-Hernickit kills sound quality.
02:40.37e-HernickAll my sound recordings are in .wav
02:40.44mog_homeas my ears suck
02:40.44JunK-Ye-Hernick: i like gsm.
02:40.47e-Hernickgsm is evil.
02:40.51jcohenits the built in demo file, I'm just trying to get it to play a demo
02:40.56DarthCluee-Hernick, it sounds like you are doing it the right way by detecting the callerid and not transferring if it the matches the number you are trying to transfer to.
02:41.01e-HernickIt introduces horrible compression glitches
02:41.12DarthCluejcohen, you have it as demo-congrats.gsm, you need to remove the .gsm
02:41.16salviadudis mp3 > gsm?
02:41.18e-HernickDarthClue, sure, it works. But I have no way to tell the originating context what's happening.
02:41.21DarthCluegsm works well enough
02:41.22JunK-Ye-Hernick: im exactly like mog_home, i see no difference.
02:41.24mog_homemp3 is different than gsm
02:41.27e-Hernickmp3 != gsm
02:41.39mog_homemp3 for music on hold in asterisk
02:41.41mog_homenothing else
02:41.50mog_homei use gsm everywhere as it is small and cheap as free
02:41.51JunK-Ymog_home: sure, for my ipod :)
02:42.04e-Hernickyou must be recording with 20$ microphones then
02:42.42JunK-Ye-Hernick: who cares, im a dev, not a sound specialist, i see no diff with gsm, like all my co-workers.
02:43.05MicC_damn if I can get that "Messages" button on the IP501 to do anything but dial its own extension.
02:43.05JunK-Ymog_home: i wonder if i should buy the ipod nano.
02:43.10MicC_I is confused
02:43.12mog_homeits pm;y isb
02:43.16mog_homeits only usb *
02:43.20JunK-Yyo Nugget.
02:43.33e-HernickWell, you might not hear the difference, but your users will.
02:43.33mog_homeso i wont use simply because i dont have usb2 but i do have firewire stuff
02:43.41jcohenthanks all for the help
02:43.49e-HernickAt least on the subconscious level. By dropping awful compression, you improve the quality of your asterisk system.
02:43.52DarthCluejcohen, that fix it?
02:43.55jcohenyep
02:44.08jcohennow off to learn more about asterisk
02:44.15JunK-Yive firewire too, apparently, all the ipod works with the usb2, even if they're really old (like mine)
02:44.18mog_homeonly bills, telemarketrs and family call, so better let them cringe
02:44.27e-HernickOf course, if you make your recordings with awful 20$ microphones or worse, telephones, it won't make much of a difference.
02:44.29jcohenlol
02:44.34JunK-YDarthClue: the diamonds guy!
02:44.36JunK-Ysup^
02:44.53DarthClueJunK-Y, i had nothing to do with that.  I was just enjoying the cab ride.
02:44.59*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
02:45.00mog_homeashe is bored and avoiding school work
02:45.08mog_homeerr as he
02:45.12jcohenoh I got a question
02:45.17jcohenwhat is the _X. line about?
02:45.19mog_homesoot
02:45.19miahif im getting a "we're sorry this call cannot be completed at this time" response when i dial, is that a issue with my voip provider?
02:45.23mog_homeits a regex
02:45.29mog_homeit matches any number longer than a digit
02:45.36DarthClueit matches anything starting with a digit
02:45.46DarthClueer, or maybe what mog said.
02:46.00DarthCluemiah, could be.
02:46.06mog_homeit will match a single digit, but it has to time out
02:46.10mog_homeif im not mistaken
02:46.13mog_homeand i could be
02:46.14twistedokay
02:46.19mog_homeas i use _. because im lazy
02:46.20twistedDEBATE OF THE EVENING!!
02:46.23jcohenbut I thought the s, was suppose to it
02:46.32filehi twisted
02:46.36mog_homes is for matching things with out an extension
02:46.41mog_homelike an analog phone call
02:46.42jcohenahhh
02:46.42mog_homecomming in
02:46.47mog_homeits the start extension
02:46.48DarthCluejcohen, well, it doesn't match on numeric entries
02:46.50twistedDO I, 1) tear down my current pbx and turn it into a media server, or 2) try and use that HUGE hunk of machinery in the other room as the media server instead
02:46.59DarthClueer, yeah, what mog said
02:47.01twistedsure, the machine in #2 is MUCH faster, etc.
02:47.12mog_home#2 twisted
02:47.13twistedbut the pbx is currently running on a SFF mini-itx box.
02:47.21DarthCluetwisted, use #2
02:47.21mog_homeas no one calls you anyways ^_*
02:47.35twistedDarthClue, sad thing is, i'm leaning that direction
02:47.42twistedbut it wouldn't make a very nice looking set top box
02:47.47pbdtwisted: What does a PBX need anyway? Wet string and a p-II.  I'd go with option 2)
02:47.48miahmy provider is broadvoice, been having issues getting my dial plan to allow me to dial out.. such a frustration
02:47.51jcohenlol
02:47.57jcohenbe back in a bit
02:48.06mog_homewhats the problem miah
02:48.07twistedwell, the thing is
02:48.18twistedthe machine in #1 is fanless, small, and pretty
02:48.31DarthCluetwisted, it doesn't need to sit on the set, just needs cables long enough with enough shielding to attach it to the set
02:48.39twistedDarthClue, i don't live in the ghetto.
02:48.51pbdmah: That message could mean many things- provider issues, yes.. it can also mean that you don't have a match in your incoming context for the extension being sent in.
02:48.54DarthCluetwisted, nobody said you did
02:49.08twistedDarthClue, ie, i don't want wires running all over kingdom kum
02:49.26twistedBUT, i don't wanna tear down the pbx and rebuild it either
02:49.29twistedgrrr.
02:49.30DarthCluetwisted, apartment?  yeah that blows.  house?  put em in the walls.
02:49.38twistedyeah, apartment
02:49.45twistedif it was a hosue
02:49.46twisteder house
02:49.50twistedi'd build the damn pc into the wall
02:49.53DarthCluewireless media transmitter maybe?
02:49.53pbdtiwsted: So run the big peice of machinery in the other room as the AV server *and* use the mini-itx as a head. :)
02:50.20twistedpbd, nice idea, but i'm talking abotu windows xp media edition
02:50.34twistedor whatever it is
02:50.34mog_homeEwWWWWWWWWWWWWWWW
02:50.37mog_homemythtv
02:50.39pbdAre you daft?  That's what MythTV is for.
02:50.44twistedmog_home, before you start with your mythtv propoganda
02:50.49twistedi have had shitty luck with myth
02:50.52twistedHORRIBLE
02:51.00mog_homeill hook you up
02:51.03mog_homefor a price.....
02:51.03pbdMedia Center runs *hot*.. that mini-ITX may not be up to it.
02:51.22miahhrm i think it is broadvoice
02:51.26twistedpbd, i've seen smaller enclosures w/o fans withstand it
02:51.35twistedbut i dono
02:52.05pbdtiwsted: I've seen machines without monitors running Windows XP. I like it better that way, personally. :)
02:52.10twistedi need to get a tuner card that can do digital anyway
02:52.16twistedpbd, yeah, me too.
02:53.16twistedmeh
02:55.16*** join/#asterisk Moc_ (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
02:56.24spacklemythtv rocks.... almost
02:57.29Nuggetthat's the story of open source. :)
02:57.55mog_homebah
02:58.03mog_homeopen source rocks, unless your lazy
02:58.06mog_homethen you have to pay
02:58.07MicC_hey spackle
02:58.23MicC_no luck with that "Messages" button on the IP501.
02:58.26MicC_weird eh?
02:59.14spackleMicC, yeah did you find the item in the config where its set?
02:59.17pbdG'night all.
02:59.29MicC_yeah
02:59.32MicC_its either 1 or 0
02:59.33MicC_lol
02:59.50MicC_there must be somewhere else to set the extension or number to dial
03:00.02spackleMicC, there is more to it thant that - yeah, it's a little non-intuitive.
03:00.11spackleMicC, give me a second
03:00.14FuriousGeorgewhen your lookint at the back of the tdm, the jack closest to the mb is bay #1 right?
03:00.24Moc_I can give you my ip500 msg settings if you want
03:00.29Moc_it only 2 settings to set
03:00.36MicC_I found one...where is the second?
03:00.50miahanybody else using broadvoice and having issues right now?
03:00.53*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
03:01.04thisbox71is anyone able to do conference calling with asterisk (not meetme channels)
03:01.06Moc_<PROTECTED>
03:01.21Moc_you need to set callBackMode at contact, and replace 855 with your voicemail extention
03:01.31Moc_that it
03:01.42MicC_ok...thats cryptic.
03:01.43MicC_lol
03:01.43spackleThanks Moc.  It is a little non-intuitive yes?
03:01.55Moc_not once you know it ;)
03:02.23Moc_polycom phone arnt toys, that is what I love about there, there is so much options on them they are crazy..
03:02.47Moc_the wiki is a good place for help on * specific settings
03:02.53Moc_www.voip-info.org
03:02.53spacklethat's the truth.  Did you see the new 601 + sidecar modules for it?
03:03.02Moc_sidecar ? nope
03:03.19Moc_Oh nice !!!
03:03.38Moc_exactly what was missing from polycom hehe
03:03.56spacklesome people would say a backlight too
03:03.57Qwellsip sidecar?
03:04.01Moc_kind of suck that I'll have to spend money on that !!!
03:04.10Moc_yes backlight would be amazing
03:04.10Qwellspackle: and quick boot times. ;]
03:04.20Moc_ok quick boot times also
03:04.24Moc_but still hehe
03:04.33Moc_you dont reboot your phone everyday..
03:04.47spackleMicC, got any other questions while Moc is about?
03:04.59Moc_;)
03:05.01spackleMicC had to do a speedy installation of a bunch of phones.
03:05.15miahhrm weird looks like the first few digits are getting trimmed off my phone number when i dial out
03:05.27*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
03:05.41miahrather than 617459nnnn i se sip:17459nnnn@sip.broadvo
03:05.47spackleMoc_ how do you have your dial plan in your Polycom?
03:05.51miahso the 16 is being removed by something..
03:06.18miahah n/m i know whats doing that
03:06.37Moc_[2-9]11|0|9011xxx.T|9[0-1][2-9]xxxxxxxxx|9[2-9]xxxxxxxxx|[1-8]xx
03:06.43miahok, back to 'you are not able to make this call please contact your system...
03:06.53Moc_that it
03:06.57mog_homeDarthClue: put the book down and put your hands on your head
03:07.52*** join/#asterisk rnasby (n=rnasby@adsl-68-20-20-198.dsl.chcgil.ameritech.net)
03:07.53spackleI'm stealing it.
03:07.56mswMoc_: the default digitmap on the polycoms is pretty useless
03:08.07mog_homemog does not squish
03:08.08miahgrrrr
03:08.12miahwtf is stopping me from dialing out
03:08.12spacklethe default plan makes me hurt
03:08.46Moc_mine work for 9 for outgoing in Can/US and international, also include everything else for my local dialling using 3 digit
03:15.36spackle~weather kdsm
03:16.29spackleMoc_: so what's the new job?
03:16.53*** join/#asterisk santiago (n=santiago@63.245.86.145)
03:17.36Sedorox~weather ipt
03:18.42FuriousGeorgeim using gentoo, and on this one box starting asterisk with the init scrip doesnt work.  i gotta launch it by hand.  the script itself might as well be greek to me, and it never complains when it starts up
03:18.42DarthClue~weather kotv
03:18.44MicC_Moc/Spackle: that did the trick...thanks guys
03:18.50DarthClue~weather ktul
03:18.55Moc_spackle, doing good extreamly busy..
03:18.59Sedorox~weather kipt
03:19.14spackleMoc_: not at a law firm though?
03:19.15FuriousGeorgei check messages all it says is asterisk already started.  anyone have experience with that
03:19.18Moc_nope
03:19.42Moc_they are just too slow to move to new tech
03:20.14spackleMoc, Any opensource or linux at the new job?
03:20.20Moc_oh yea..
03:20.28Moc_it only Linux servers...
03:20.29miahhrm
03:20.31Moc_Run Asterisk
03:20.53spackleMoc_ will they give you time to work on Asterisk on the job?
03:20.58Moc_I have setup asterisk there for about 8 month now
03:20.58*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:21.01miahi keep seeing "Outbound Registration: Expiry for sip.broadvoice.com is 20 sec (Scheduling reregistration in 15999 ms)
03:21.05Moc_spackle, Im supose too..
03:21.13miahi googled for that and it looks like this issue was fixed like 1 year ago
03:21.14miahso wtf
03:21.22spackleMoc_ that's excellent?
03:21.24Moc_Just rightnow, they are fucked in another project, and I gota help them
03:21.39miahi'm running 1.0.8
03:21.54Moc_but once this is done, I'll put more time... and more time this project will need
03:22.52spacklecool, that's great news.
03:23.50DarthCluemiah, 1.0.8 is really, really old
03:24.09Nivexheck I'm still running 1.0.7 packaged in Debian :)
03:24.21Qwell1.x?  psh
03:24.28Qwell0.97, baby
03:24.34DarthClueNivex, I sure hope you don't need help cause we don't support that.
03:24.37miahhah
03:24.49DarthClueQwell, don't make me hurt you.  I know how to get what file to do what I want.
03:24.53NivexDarthClue: nah, it's still runnin' quite nice for what I do.  If it breaks, I'll bump it to latest :)
03:24.57miahi know its fairly new, what im saying is if this issue was around 1 year ago why am i hitting it =) its obviously been fixed
03:25.29DarthCluemiah, not in 1.0.8 since 1.0.8 is way over a year old if I remember right.
03:25.47QwellDarthClue: only a few months
03:25.51DarthClueand I think 1.0.8 was really messed up too.
03:26.06UberbotDoes anyone know if it is possible to play a message to a caller AND dial an extension at the same time?
03:26.22DarthClueQwell, it's late, I don't really care, just trying to nicely suggest that he should run HEAD
03:26.23miahhrm, its the newest stable release in gentoo
03:26.40Qwellmiah: thats why we dont use packages...
03:26.40miahi could try a newer version
03:26.42NivexI've heard evil things about the gentoo ebuild
03:27.32DarthCluewell, I have to get up early so I'm gone.  Just remember that file is in charge, for whatever it's worth.
03:28.00spackleDon't you need to tuck file in before you go?
03:28.10miahi'll try 1.0.9
03:30.17spackleMoc_ are you still here?
03:30.30Moc_yes
03:31.24spackleMoc_ have you done any soft buttons on Polycoms?  somebody was asking today if they could add more choices than just the 3 or for buttons at the bottom of the screen
03:31.38spackleI was guessing it was possible
03:32.15Moc_add more than 3 line ?
03:32.48spackleIt sounded like they wanted more button options for features.
03:33.09Moc_no idea.. I know you can reassign all the buttons
03:33.25spackleI was thinking you could add menus to the buttons and sub-options to each menu.
03:33.43Moc_no idea about that
03:34.35spackleMoc_ It was suggested that the microbrowser on the 600 could do it.  I dunno, just thought I would ask you.
03:35.43Moc_I do not know..
03:35.53Moc_I keep the standards settings (easier to manage)
03:35.55BhaalWKI have 2 asterisk servers linked .. Both behind firewalls, serverA has port forwarding on for iax2 to get through, serverB goes out through the firewall and connects to serverA and authenticates.  I can route all calls from sip fones connected to serverB out through serverA .. But how do extensions on serverA call extensions on serverB when serverA doesnt auth to serverB?
03:36.00Moc_and never required more
03:36.16fileZX81 just told me 3 ways to kill a person
03:36.37salviadudi think zx speaks german
03:36.40fileVERY interesting
03:36.45filehe's from New Zealand
03:37.05salviadudhe gave me a call on FWD
03:37.05spackleMoc, Agreed, standards and plain vanilla where possible.
03:37.08salviadudnice guy
03:37.25fileI enjoy talking to him
03:37.35salviadudfile you got fwd?
03:37.57filenot that you can reach on
03:38.07salviadudreally, why?
03:38.12fileit goes to a conference room
03:38.17salviadudohhh
03:38.21tessierfile: I'll tell you 4: Shoot the in the head. Stab them with a knife. Poison them with cyanide. Hang them with a rope.
03:38.31hardwireblah
03:39.27Vcoyou forgot "patch windows 200 pro rtm to current without windows update"
03:39.37filesalviadud: 61140 will call me now
03:39.43fileHAHA
03:40.35NivexI think that guy had me confused with zx81 somehow (zx sounds like nivex?).  I mentioned I spoke German on the phone to him.
03:41.00fileah
03:41.41Nivexnow mind you, I don't speak German all that great, but I only know a few words in Spanish.
03:41.49miahgrr, still have the same issue with 1.0.9
03:44.36QwellI need a grammar check
03:44.45Qwell"he gave me the answers I was looking for"  doesn't sound right
03:45.17FuriousGeorgeso bay 1 on the tdm must be the jack farthest from the mb, right?
03:45.42spackleFuriousGeorge, port one, yes.
03:46.11*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
03:49.27hardwirehey
03:49.29hardwireheads up
03:49.31hardwireanybody like
03:49.35hardwirehave mplayer installed?
03:50.33spacklehardwire, what do you mean?
03:50.45hardwiredo you know what mplayer is?
03:51.29spackleIs it a mp3 player or an X app for all media?
03:52.04miahmplayer plays just about any video
03:52.17SplasPoodHow much did you wager?
03:52.19Vcodiarrhea
03:52.23Vcoi'm sorry..
03:52.25spacklek, then not on asterisk.
03:52.28Vco"what is diarrhea...?"
03:52.45spackle~diarrhea
03:52.51spackleheh
03:52.55Vcoheh..h.eh.
03:53.06spacklejbot?
03:53.08SplasPoodWhen you're slidin into third, and you...
03:53.20spackleGeez.  2nd grade
03:53.22hardwireah
03:53.23hardwirewell
03:53.28hardwireI have a live theora/vorbis stream
03:53.36hardwirewhich is looking at our busy intersection
03:53.52spackleI have VLC, bet it could play it.  What's the address?
03:53.54*** join/#asterisk bmg505 (n=leon@rndf-146-30-131.telkomadsl.co.za)
03:54.19hardwireVLC can
03:59.39enderhrm, I need some help w/ setting caller ID stuff.  Anybody familiar w/ this?
04:03.29UberbotDoes anyone know if it is possible to play a message to a caller AND dial an extension at the same time?
04:05.20enderbackground, wait, dial
04:05.51*** join/#asterisk trig_hm (n=jb@home.monkeypr0n.org)
04:06.31spacklelooks green now.
04:06.39Corydon76-homeUberbot: no, it's not possible
04:06.56*** part/#asterisk mstocco (n=mario@207.212.29.195)
04:07.03Corydon76-homeUberbot: you could play music on hold while dialling an extension, though
04:08.16enderwhy couldn't you s,1,Background(message); s,2,Wait(50; s,3,Dial(foo)  ?
04:08.43endermost likely the dial will kill the background message.
04:09.11Corydon76-homeBackground doesn't play the sound in the background while continuing
04:09.29enderah, I see.
04:09.39Corydon76-homeBackground plays the message while listening for DTMF.  Compare Background to Playback, which does not listen for DTMF
04:09.48enderright.
04:10.00enderwhat happens when background finishes playing, just goes to the next priority right?
04:10.02Corydon76-homePlaytones will do that, however
04:10.05*** join/#asterisk denon (i=denon@synapse.subneural.net)
04:10.05*** mode/#asterisk [+o denon] by ChanServ
04:10.08Corydon76-homeender: correct
04:10.24Corydon76-homeHowever, Playtones are just that -- tones, not a sound file
04:14.20enderCorydon76-home: ever played w/ setting CallerID for calls coming in?
04:14.32Corydon76-homeYep
04:14.39enderCorydon76-home: I want to set a specific caller ID for all calls coming in via an iax2 channel
04:14.51Corydon76-homeMostly to work around single-line-display callerid on certain SIP phones
04:14.53enderright now I'm getting Unknown for caller ID
04:15.17enderon the remote system where I dial an IAX2 I have this:
04:15.31enderexten => _5XXX,1,SetVar(CALLERIDNAME=Outside Caller)
04:15.32enderexten => _5XXX,2,Dial(${ASTERISK}/${EXTEN});
04:15.52Corydon76-homeWhy not just set a callerid in iax.conf for that user?
04:16.04UberbotCorydon76-home, while the MOH was playing, could the caller dial another extension?
04:16.04ender${ASTERISK} happens to be a variable for the iax2 channel.
04:16.38enderCorydon76-home: eventually the remote * box will have both a PRI and a T1 line to an old PBX>  The PRI line should have valid caller ID, so I don't want to override that right now.
04:17.00Corydon76-homeUberbot: um, I was envisioning moh while the extension was ringing
04:17.19enderCorydon76-home: can you set caller ID on say an entire zap channel?
04:17.33UberbotYes, so am I, but the caller may want to bail out and call another extension.
04:17.50Corydon76-homeUberbot: no, not in that situation
04:17.57UberbotDidn't think so.
04:18.11Corydon76-homeThe caller can only type another extension while an application is listening for DTMF
04:18.29UberbotIt's a shame that dial doesn't listen.....
04:18.48Corydon76-homeWhy?
04:19.28Corydon76-homeender: yes, just set it in zapata.conf before you declare that channel
04:19.42Corydon76-homeender: to reset, set callerid=asreceived
04:19.43enderCorydon76-home: yeah, found that, thanks.
04:19.50UberbotI've got it so that incoming callers get a message indicating that # to go to voicemail, * to try to ring the phones.  or any other extension.  15 second timeout to voicemail.
04:20.20UberbotMy wife doesn't want to "inconvenience" callers by making them press * to ring our phones.....
04:20.37UberbotI, on the other hand, want to be able to dial in and call other extensions.
04:20.40Corydon76-homeSo once they make their decision, then you take action.  What's wrong with that?
04:20.50UberbotI LIKE IT,  She doesn't.
04:20.52enderwell thats poopie.
04:21.07UberbotYup.  :-D
04:21.15enderNoOp(${CALLERID}) shows the right caller ID, but my phone still shows Unknown/Unknown
04:21.15Corydon76-homeUberbot: well, first you need to agree with her how you want the system
04:21.27UberbotI was hoping to compromise.
04:21.56UberbotShe just wants a simple answering machine, it seems.
04:21.57Corydon76-homeUberbot: learn the magical words "I don't know how to make it do that"
04:22.25Uberbot"I don't know how to make it do that, but I'll check the experts on IRC."
04:22.31Uberbot:-D
04:22.32Corydon76-homeender: this on analog or PRI lines?
04:22.52enderCorydon76-home: the phone is a SIP phone.
04:23.16UberbotI've even considered having it configurable based on CID of who's calling.....
04:25.00UberbotDo you see any other posibilities?
04:25.46enderCorydon76-home: call originates on a T1 line (not PRI), I set the CallerID because it doesn't exist at that point.  I then pass the call along an IAX2 channel to another * box which then dials the SIP phone.
04:26.58Corydon76-homeSo the callerid is not being trusted...
04:27.56endernot from calls on the T1 line.
04:28.11enderthere is no caller ID informatino.  The PBX the T1 is connected to is not capable of passing callerID through.
04:28.25ender<PROTECTED>
04:28.25ender<PROTECTED>
04:28.40enderwhy wouldn't the phone see Outside Caller on it instead of Unknown ?
04:29.11Corydon76-homebecause it's not being passed through IAX
04:29.32enderno, it is.
04:29.43enderthose two lines are from the system the SIP phone is attached to
04:29.57enderthats AFTER the IAX link
04:33.04Corydon76-hometrustrpid=yes in sip.conf ?
04:33.40Corydon76-homeand restrictcid=no ?
04:33.47enderneither.
04:34.30Corydon76-homeDon't know, then
04:36.25*** join/#asterisk _omer (i=o@203.215.180.250)
04:37.20*** part/#asterisk _omer (i=o@203.215.180.250)
04:37.39*** join/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com)
04:38.07*** part/#asterisk Samoied (n=Samoied@200.101.233.182)
04:38.17enderCorydon76-home: I tried trustprid=yes and that didn't help.
04:38.32Corydon76-homeor trustrpid?
04:38.48enderyeah
04:38.51Corydon76-homeHelps to spell it correctly
04:39.00endertrustrpid=yes
04:39.15endertypod it when typing here in irc, the config file had it right.
04:39.32Corydon76-homek
04:39.55Corydon76-homereally don't know, and I'm about to fall over at the keyboard, so I'm going to bed
04:40.51enderlater.
04:48.21UberbotThe wiki mentions bristuff.  Will the apps in this patch work with a zap channel?
04:50.00*** join/#asterisk jadeisfalling (n=jadeisfa@ip68-106-235-243.ph.ph.cox.net)
04:50.00FuriousGeorgeso that presence thats in cvs head...  is that modular or do i need to install all of cvs head to use it?
04:51.47*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:52.36*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
04:56.30*** part/#asterisk jadeisfalling (n=jadeisfa@ip68-106-235-243.ph.ph.cox.net)
05:01.57mostyi have a pap2, and i have asterisk setup with two identical sip accounts (one for each port on the pap2), but i can only get one of those accounts to register, what could be wrong?
05:02.50mostyasterisk says "registration failed", how can i find out why it failed?
05:04.23niZonincorrect password?
05:05.13swm_debug it :)
05:05.23mostyoh wait, i know what it is, i copied the md5secret thinking that would mean an identical password, but it doesn't
05:05.36swm_LOL
05:05.53swm_~beat mosty
05:05.55jbotACTION beats mosty with a large stick.
05:05.57*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
05:06.57niZon~beat off
05:06.58jbotACTION beats off with a large stick.
05:07.04Qwellthats awesome...
05:07.07niZonsorry, had to
05:07.16QwellL3 employees in Anaheim had to be evactuated today, due to rain
05:07.23Qwellevacuated*
05:07.28niZonlol
05:07.37niZondid they forget to build a roof?
05:07.43Qwellit was caving in, heh
05:08.30niZoncrappy building
05:09.14FileParadox~spank off
05:09.16jbotACTION bends off over his knee and tatoos 'ibot' on off's pasty white buttocks.
05:09.36FileParadox~spank his dick
05:09.37jbotACTION bends his dick over his knee and tatoos 'ibot' on his dick's pasty white buttocks.
05:09.56jcohenif I change the extensions.conf file can I have asterisk reload it with out disconnecting anyone or shutting asterisk down?
05:10.06FileParadox~whoami
05:10.08jbotyou are fileparadox, or n=admin@digitaldatabits.net on #asterisk and it's Wed Sep 21 05:10:08 2005. Don't believe me? Ask otmar!
05:10.11niZonlol
05:10.28niZonjcohen: reload won't disconnect anyone
05:10.32FileParadox~whack
05:10.38jcohenis that all I need to do?
05:10.51niZonasterisk -r -x 'reload' will do it :P
05:10.51FileParadox~love
05:10.53jbotIf you love <insert item> so much, why don't you marry it
05:10.55jcohenthanks
05:11.10niZon~love asterisk
05:11.11jbotIf you love asterisk so much, why don't you marry it? (oooooh)
05:11.19FileParadox~love bkw_
05:11.20jbotIf you love bkw_ so much, why don't you marry it? (oooooh)
05:11.25*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
05:11.29jcohen~love jbot
05:11.31jbotIf you love jbot so much, why don't you marry it? (oooooh)
05:11.31niZon~love my rectum
05:11.32jbotIf you love my rectum so much, why don't you marry it? (oooooh)
05:11.47jcohenthanks
05:11.47FileParadox~kill my dick
05:11.48jbotACTION shoots a hyper-charged  positrino gun at my dick
05:12.08FileParadox~kill his dick
05:12.09jbotACTION shoots a magneto-ionized anti-graviton gun at his dick
05:12.26jcohen~help
05:12.37FileParadox~reboot
05:12.39jbotNot on your life cowboy :(, or are you using Window$?
05:12.39FileParadox~restart
05:12.40jbotthis aint no stinkin' winblows, or "Microsoft Windows: Your mouse has moved; please restart for changes to take effect"
05:12.46FileParadox~fart jcohen
05:12.47jbotACTION farts in jcohen's general direction
05:12.57*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
05:13.01niZon~help
05:13.07FileParadox~hack
05:13.09niZonaw crap
05:13.15[hC]Hi guys
05:13.18FileParadox~commands
05:13.45FileParadox~jbot help
05:13.47[hC]Im having occasional problems where i get one-way audio dropouts for about 5-6 seconds at a time, sometimes two-way audio loss, but only for a few seconds
05:13.54[hC]i cant for the life of me narrow down whats happening
05:14.21niZonbad cable?
05:14.38FileParadox~lobotomy niZon
05:14.40jbotACTION pulls out a rusty saw to perform a lobotomy on niZon
05:14.46niZonnoooo
05:14.49[hC]Im mainly trying to figure out how to debug it
05:15.02FileParadox~news
05:15.14FileParadox~news add niZon Sucks
05:15.37niZon~md5 blah blah blah blah
05:15.38FileParadox~news text 1 Something new
05:15.46FileParadox~news
05:16.22FileParadox~piglatin Hello everyone how are you today
05:17.07niZon~piglatin Hello there, I pwn teh noobz
05:17.29FileParadox~unlobotomy jbot
05:17.50FileParadox~freshmeat
05:17.52jboti heard freshmeat is at http://www.freshmeat.net
05:18.06niZonwhere is freshmeat
05:18.07FileParadox~ircstats
05:18.07jbotCurrently I'm hooked up to irc.us.freenode.net:6667 but only for 12d 14h 56m 34s.  I had to reconnect 4 times.   Connectivity: 100.00 %
05:19.03FileParadox~rot13
05:19.13QwellFileParadox: message him.  That is incredibly annoying
05:19.27FileParadox~rot13 give them hell
05:19.27jbottvir gurz uryy
05:20.14*** join/#asterisk MGSsancho (n=user@adsl-67-127-235-22.dsl.irvnca.pacbell.net)
05:20.41*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
05:20.57*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-74.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
05:23.13niZon~rot13 rot thirteen in hell
05:23.13jbotebg guvegrra va uryy
05:23.29Qwellno, no, no
05:23.32Qwell~rot13 1337
05:23.32jbot1337
05:23.43niZonlol
05:23.50niZon~rot13 94325904852384528349804385
05:23.50jbot94325904852384528349804385
05:24.00*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-74.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
05:24.09enderniZon: please use /msg jbot.  Thanks.
05:24.19niZonbah
05:24.21niZoneveryone's doing it
05:25.41swm_Yeah everyone's doing it
05:26.18*** join/#asterisk many (n=many@daheim.ukeer.de)
05:33.49mostyi'm having trouble making sip -> sip calls via a public asterisk gateway. both sip clients are behind NAT. when the call is connected, neither party can hear the other. the asterisk machine has a firewall rule that allows all rtp traffic to the NAT'ed sip clients. what else should i check?
05:35.49mostyalso, sip -> asterisk -> isdn -> landline calls work fine
05:36.55*** join/#asterisk litage (n=nick@203.201.99.69)
05:41.54FuriousGeorgeany softphones able to alert the user when there is a new vm?
05:42.28FuriousGeorgeturn on a little light or something
05:43.43*** join/#asterisk clive- (n=pirch@ndn-165-157-14.telkomadsl.co.za)
05:44.35FuriousGeorgewould be spectacular if there were some way to make it work with eyebeam but i dont even know if the client supports it.  if one were to get asterisk messaging working perhaps it could be used to alert user of a new vm.
05:46.05enderemail doesn't suffice?
05:46.21endermy clients get alerted of new vm by email as well as WMI on they're hardphones.
05:46.27twistedFuriousGeorge, even x-lite has vm indicator
05:46.32twistedyou just have to turn it on in sip.conf
05:46.44FuriousGeorgetwisted: cool
05:46.50Qwellokay...anybody in southern california happen to be around?
05:46.57FuriousGeorgeender: the people i deal with at work hate any mail that doesnt have a stamp
05:46.59Qwelldoes it stink like garbage outside?
05:47.05FuriousGeorgelol
05:47.08enderFuriousGeorge: wuf.  I'm sorry
05:47.13twistedQwell, lol
05:47.14QwellI'm serious...
05:47.19Qwellcompletely serious, in factr
05:47.23twistedQwell, did you poop in the street?
05:47.24enderFuriousGeorge: so attach a stamp gif to all t heir email (:
05:47.29FuriousGeorgelol
05:47.42Qwellsomebody in socal, go outside, and smell for me, would ya? :p
05:47.46endertwisted: care to poke at a Polycom callerID problem?
05:47.51clive-would anyone have an idea why bandwidth usage of iax2 trunking on the receiving side is using much more bandwidth than the transmitting side?
05:47.57FuriousGeorgesocal fits 10 new jerseys
05:48.09FuriousGeorgeanyone in pensylvania seen my keys?
05:48.11twistedender, as long as it's quick
05:48.17FuriousGeorgehowabout you RI
05:48.47endertwisted: well, NoOp right before the Dial line shows $CALLERID as 'Outside Caller'.  However the Polycom dialed shows 'Unknown' for caller ID.
05:48.51endertwisted: any thoughts?
05:49.11twistedender, no number?
05:49.13Piranha-so no amd64 support for asterisk yet eh?
05:49.30*** join/#asterisk santiago (n=santiago@63.245.86.145)
05:49.38endertwisted: I'm not assigning a number w/ my SetVar.  No number comes through the source, so I'm just setting it to name 'Outside Caller'
05:49.48twistedyou're using setvar?
05:49.48enderPiranha-: I'm running asterisk on amd64s in 64bit mode...
05:49.55Piranha-what os
05:49.56endertwisted: yah
05:50.00enderPiranha-: CentOS4 x86_^4
05:50.01Piranha-freebsd wont compile it
05:50.02twistedender, that's why it's not working.
05:50.05Piranha-nor the port
05:50.19endertwisted: ah ok.  What would I need to do in order to get the Polycom to show it?
05:50.20Piranha-the port is i386/sparc64 only.. and I get errors in compiling myself
05:50.31enderPiranha-: sorry, I'm Linux, !BSD
05:50.38twistedender, on 1.0.x, SetcallerID('whatever')
05:50.41Piranha-hehe I figured :)
05:50.49twistedender, on CVS HEAD, SET(CALLERID="wahtever')
05:51.00enderti's 1.2.0beta1
05:51.06endertwisted: actually wait.
05:51.07*** part/#asterisk santiago (n=santiago@63.245.86.145)
05:51.08*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-74.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
05:51.14Qwellso, no, seriously...no takers?
05:51.15endertwisted: I'm not using SetVar
05:51.23twistedender, ok, figure out what you're using
05:51.44twistedQwell, it smells like ass outside
05:51.44endertwisted: on a remote * box, in zaptel.conf, I have 'callerid='Outside Caller''
05:51.59endertwisted: this is to set it on all calls coming from our Zap T1
05:52.16twistedQwell, oh wait, that's just my donkey.
05:52.29twistedQwell, call the local news
05:52.33*** join/#asterisk [MUPPETS]Gonzo (i=gonzo@80.69.47.16)
05:52.34[MUPPETS]Gonzo...
05:52.36[hC]hey so... im having some interesting problems where i have one way, sometimes both ways, audio dropouts for 5-10 seconds at a time, then the call returns. what is the most comomn cause of this, or how might i debug it?
05:52.48JerJerpacket loss
05:52.49twistedender, k.. does it work on other phones?
05:52.56endertwisted: I have no other phones to try.
05:52.57[hC]ive seen it happen on multiple iax connections, as well a simple sip->*->sip between two phones
05:53.02endertwisted: only Polycom 501s and 301s
05:53.26twistedender, what protocol are you using to send the calls to the second box?
05:53.32enderiax2
05:53.42twistedare you AUTHENTICATING iax2?
05:53.47endertwisted: yes
05:53.48twistedor just passing them as guest?
05:53.53enderauthenticated.
05:53.56twistedk
05:54.12twisteddoes sip debug show the callerid correctly in the message sent to the phone?
05:54.19endertwisted: like I said, on the box w/ the sip phones, a NoOp shows the $CALLERID to be Outside Caller
05:54.23enderlets try.
05:54.42twistedyou try.  I'll wait
05:55.13endertwisted: ah, lookst like no.
05:55.26enderINVITE sip:5713@10.0.2.240:5060 SIP/2.0
05:55.26enderVia: SIP/2.0/UDP 10.0.2.1:5060;branch=z9hG4bK51df2d26
05:55.26enderFrom: "Unknown" <sip:Unknown@10.0.2.1>;tag=as30bf9eae
05:55.27twistedwhat is it saying in From:
05:55.30twistedahhhhh
05:55.38*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
05:56.00enderstrange.  I thought that pulled from CALLERID.
05:56.09twistedwell, sorta.
05:56.13ender10.0.2.1 is the server itself.
05:56.19twistedIIRC, in the code, we check to see if we have a number first
05:56.26ender<PROTECTED>
05:56.26ender<PROTECTED>
05:56.29twistedif we don't, we set the number to unknown
05:56.47enderthats a NoOp(${CALLERID})
05:56.49*** join/#asterisk bumblefsck (n=bumblefs@69-160-158-193.ontrca.adelphia.net)
05:56.55twistedthen, we duplicate the number into the name
05:57.00twistedi understand that
05:57.07enderok, it's because no number is set?
05:57.11twistedpossibly
05:57.13twistedset a number
05:57.16enderok.
05:57.17*** join/#asterisk akrall (i=user@201.144.59.221)
05:57.20twistedeven if it's 0
05:57.40twistedif that works, i'll try a small fix, and get you to try it
05:57.42enderwould 'callerid="Outside Caller <0>"  suffice in zapta.conf ?
05:57.49twistedit SHOULD
05:57.50twistedyes
05:57.56akrallguys.. anybody using sipuras 841 with asterisk? did you have sound issues?
05:57.58Qwelldoes callerid need quotes?
05:58.09twistedQwell, nah, but we discard them anyway
05:58.13Qwellahh, okay
05:58.28endertwisted: waiting for zap to be able to take calls again....
05:58.28Qwellall quotes, or just first and last?
05:58.47twistedhuh?
05:58.51enderah.
05:58.56endernow I get '0' on the phone.
05:59.07twistedwhat does sip debug say?
05:59.09enderFrom: <sip:0@10.0.2.1>;tag=as07f241a1
05:59.20ender<PROTECTED>
05:59.21twistedbetter...
05:59.23*** join/#asterisk littleball (n=littleba@bb220-255-134-66.singnet.com.sg)
05:59.36*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
05:59.54littleballhello, who can recommend a network card for me to use asterisk?
06:00.12Qwelllittleball: almost any NIC should work fine
06:00.21Qwelljust make sure it has Linux support
06:00.26enderor BSD
06:00.29djin_iblittleball, avoind onboard if you can
06:00.36djin_ib(personal experience ;)
06:00.41twistedooooh
06:00.56twistedit almost looks like it could be the calling presentation set to restricted
06:01.11enderdjin_ib: depends on the server board.  onboard e1000 is a very good chip.
06:01.30djin_ibender, no it isn't ;)
06:01.34enderdjin_ib: especially when there are 2 of them.  1u or 2u server PCI slot space is at a premium.
06:01.40*** join/#asterisk phpboy (i=shane@tbnb-165-215-98.telkomadsl.co.za)
06:01.41twistedhey ender
06:01.44twistedbefore you dial the sip phone
06:01.51phpboyhey guys, who can help me with ISDN4Linux support?
06:01.54littleballQwell and djin_ib, i have bought the Dell 2850. it has two onboard card. I bought another intel pro 1000MT external card, it still uses the same driver e1000 as the onboard card. This is the reason i want your help and find out what network card used on your system. So that i can get a new card
06:01.56twistedput in a Setcallerpres(allowed_not_screened)
06:02.02enderdjin_ib: thousands of my customers would beg to differ in they're high network usage experiences w/ my servers.
06:02.04djin_ibEnabling both cards on my Dell 2850 gives 'frame error's' on the TE410P
06:02.17endertwisted: Sure.
06:02.17akrallguys.. anybody using sipuras 841 with asterisk? did you have sound issues?
06:02.20littleballender, e1000 doesnt' work on dell 2850
06:02.24twistedender, or better yet
06:02.26twistedjust allowed
06:02.30twistedSetcallerpres(allowed)
06:02.33enderlittleball: I don't use / sell Dells
06:02.34Qwelllittleball: why not?
06:03.06littleballQwell and ender, can you help me check what network card used in your system?
06:03.10twistedender, and remove the 0 from the callerid on the remote box
06:03.17djin_iblittleball, do you plan to use Digium board(-s)
06:03.20Qwelllittleball: Whats wrong with the intel cards?
06:03.29littleballI am using TE411P card from digium
06:03.52littleballQwell, the e1000 driver has confliction with intel card
06:04.12Qwellwhat conflict?
06:04.15*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-74.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
06:04.26djin_ibirq, probably
06:04.33littleballthe intel card is always blinking (blue color). The dell support told me that it indicates someting wrong on the system although everything is work for the timebegin
06:04.40endertwisted: that seemed to do it.
06:05.04enderlittleball: Intel onboard e1000 chips on Supermicro intel systems.
06:05.09twistedender, ah hah!
06:05.18*** join/#asterisk lucasjb (i=lucas@hosted237.stabat.com)
06:05.19enderlittleball: and tg3 broadcom chips on tyan AMD64 chipsets.
06:05.21twistedender, SOMEWHERE in there the calling presentation is being set to "restricted"
06:05.24djin_iblittleball, you're not referring to the blue light with the little button next to it, are you?
06:05.25enderok.
06:05.35littleballThis is the the email from diguim support "Do you use the onboard gigabit ethernet controller? If not, you should
06:05.36littleballrmmod the e1000 kernel module and the blinking should stop."
06:05.42twistedender, what kind of interface is your zap interface/
06:05.44littleballdjin_ib, yes
06:05.52enderHAHAHAH
06:05.56enderthats the 'locate' light isn't it?
06:06.03djin_ibyeah ;)
06:06.04enderpush it to 'locate' your server on the back?
06:06.08littleballdjin_ib, it is the blue light with the little button next to it.
06:06.22ender*chortle*
06:06.23djin_iblittleball, that has little to do with your intel board.
06:06.29endertwisted: it's a T1 line to a Fujitsu PBX
06:06.36twistedender, protocol?
06:06.42enderem_w
06:06.50twistedhehehe
06:06.59twistedthere's no number reported then, right?
06:07.04enderright
06:07.05djin_iblittleball, pull out a powercord (if you have reduntant) and watch it go orange.
06:07.08enderstupid fujitsu
06:07.08twistedokay
06:07.10littleballdjin_ib, but the support of digium told me to remove the e1000 driver to remove this blinking. But e1000 is the driver of the network card
06:07.13enderwe're getting a PRI soon to bypass it.
06:07.20twistedon em_w there's no way to set the calling presentation
06:07.23enderlittleball: digium got... confused.
06:07.26twistedso i bet there's a bug in zaptel
06:07.26lucasjbHiyas, I'm interested in strategy=leastrecent in queues.conf. Does anyone know if it's possible to have Asterisk choose the agent least recently called across all queues?
06:07.32twistedlemme see if i can find it
06:07.38endertwisted: ok.
06:07.51djin_iblittleball, I think they mistaken the light with a different one.
06:08.13endertwisted: for now, is there a global way to enable calling pres?
06:08.15*** join/#asterisk grimse (n=grimse@p5481EA1A.dip.t-dialin.net)
06:08.19djin_iblittleball, of it works (no conflicts with TE4100P) go with it.
06:08.22heath__lucasjb: just make a queue that everybody is a member of and then queue those calls there
06:08.37littleballdjin_ib, what?
06:08.37littleballdjin_ib, why it is related to powercord?
06:08.54akrallguys.. anybody using sipuras 841 with asterisk? did you have sound issues? Im getting cutoff voices when calling outsides numbers using a Te110p as E1...  people tell me they hear my voice been cutoff or scratchy.. anybody had issues like this?
06:09.09twistedender, try setting usecallingpres=yes in zapata.conf
06:09.18djin_iblittleball, it isn't. It's a "something is wrong" light from your Dell, not specific Intel Network.
06:09.20endertwisted: hah, I just set that and was about to test.
06:09.24twistedender, ;)
06:09.32twistedthing is, it shouldn't default to restricted
06:09.56endertwisted: ah, that didn't do it.
06:10.02djin_iblittleball, you can also use it to 'mark' your server from one end if you need to do something on the other end.
06:10.15endertwisted: not possibly an IAX thing?
06:10.41lucasjbheath__, I'd like many members in many queues, but Asterisk should consider calls taken in any queue when deciding who's least recently taken a call. Am I making sense?
06:10.49littleballdjin_ib, sorry, i don't understand. i mean the light from orange color ->blue color ->blinking blue color one
06:10.50twistedit's doubtful, but possible
06:11.18djin_ibLittleball, youll find a similar button on the front ;) Again, if it works (no conflicts with TE411P) go with it. And regarding the light, check the manual for explaination.
06:11.25endertwisted: is that even a possible config option in IAX?
06:11.57enderhrm, wiki says no.
06:12.01heath__lucasjb: no because if the only criteria for which agent to send to is which one has been called the longest time about then you only need one queue
06:12.02littleballdjin_ib, thanks. let me check the manual
06:12.39heath__about=ago
06:12.51lucasjbheath__, there are two criteria: the member must be part of a particular queue and be least recently called across all different queues.
06:14.41heath__ok.. then let me ask you this: say we have to queues A and B and members a and b and a is a member of A and b is a member of B, and a call gets queued to A, but agent b is the leastrecent agent... well using your criteria, the call should be abandoned even though there is a member of the campaign waiting to take calls
06:15.31heath__b is leastrecent but we can't queue to him cuz he's not a member, a is a member but is not leastrecent so we can't queue to her either
06:15.38heath__doesn't make any sense at all
06:15.48*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-74.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
06:15.58endertwisted: so I just looked closer, it's a bit funny.
06:16.04endertwisted: From: "Outside Caller" <sip:asterisk@10.0.2.1>;tag=as712bc7fe
06:16.23MikeJ[Laptop]twisted is a bit funny?
06:16.24twistedyeah
06:16.25endersee the sip:asterisk@  ?  My polycom shows "Outside Caller" \n "asterisk"
06:16.35twistedender, asterisk puts asterisk as the number
06:16.38twistedif there is no number
06:16.41enderok.
06:16.44MikeJ[Laptop]sometimes timing is everything
06:16.59twistedmaybe..
06:17.09twistedwhat did you do to get it to say that?
06:17.11MikeJ[Laptop]hello all
06:17.11lucasjbheath__, OK, thinking...
06:17.46heath__i'm not ragging, just letting you know that your logic is jacked.. happens to me all the time
06:17.51twistedender, and can you do me one more favor?
06:18.10twistedender, as soon as the call comes in from the zap channel, do a noop(${CALLINGPRES})
06:18.21twistedand as soon as it comes in from the iax channel on the second box, do the same thing
06:18.30twistedwith it set like it was when it wasn't working
06:18.34endertwisted: ok.
06:18.35lucasjbheath__, Yeah, no problem, appreciate your help
06:18.42twistedi'm DYING to know where the presentation is getting set
06:18.52twistedbecause if it's where I think it is, i might be able to fix it.
06:18.54endertwisted: the 'make it work' was the enabling callerpres.  I just set that above all the Sip dials.
06:18.57enderok.
06:18.59enderjust a sec.
06:19.11twistedahh ok
06:19.21twistedthen you can leave that in
06:19.25twistedjust call that noop before you set it
06:19.26heath__maybe you mean: queue the call to the leastrecent person in that campaign, but if there's no one available, then queue it to whoever is leastrecent in all campaigns
06:19.38endertwisted: will do.
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06:21.21endertwisted: just after the IAX link:     -- Executing NoOp("IAX2/pandora@pandora-1", "67") in new stack
06:21.38littleballdjin_ib, i mean the system status indicator. pls see http://supportapj.dell.com/support/edocs/systems/pe2850/en/ug/t1390c10.htm#wp1043338
06:21.42twistedwhat about after the zap?
06:22.07endertwisted:     -- Executing NoOp("Zap/20-1", "0") in new stack
06:22.08phpboyguys, I'm trying to build isdn4linux into my kernel... but I'm not sure how to recompile a kernel for linux (gentoo) anybody got a good doc for me?
06:22.16endertwisted: I had a small error there, had to reset and run again.
06:22.19twistedender, okay, so iax is being a bitch.
06:22.25twistedlemme take a quick glance
06:22.25endertwisted: looks like it.
06:22.43lucasjbheath__, OK: say we have users 'a' and 'b'. 'a' takes calls only from queue 'A', but 'b' takes calls from queues 'A', 'B' and 'C'. 'a' has answered a call from 'A' only 7 minutes ago, 'b' has not answered a call from 'A' for 10 minutes, but has since taken a call from 'C' only 4 minutes ago. The least recent in queue 'A' is 'b', but we want Asterisk to check all queues, and therefore choose 'a' for the next call on 'A'. Make sense?
06:23.03twistedender, omg.. if i'm looking at this right, i found the problem.
06:23.12enderphpboy: why use gentoo, the rebuilders ricerocket, if you don't really know how to rebuild stuff?  Wouldn't it make more sense to use a distro more suited for you?
06:23.18endertwisted: that was rather quick.
06:23.20twistedender, and if it is, it's stupidly simple.
06:23.44endertwisted: enlighten me?  (:
06:23.57twistedif we don't recieve a caller id number
06:24.02heath__lucasjb: hold up, (drawing a diagram) :)
06:24.04twistedwe set the calling presentation to NOT AVAILABLE
06:24.05phpboyender: I'm a BSD kiddie... not Linux... so I can't do it in any distro :<
06:24.11lucasjbheath__, lol
06:24.15twistedWHICH, might be causing the whole damn issue
06:24.24enderphpboy: Try CentOS, isdn4linux is included in the stock kernel.
06:24.32endertwisted: ah HAH.
06:24.39twistedhmm
06:24.45endertwisted: what is your checking for the number look like?  Looks for something INSIDE <>, or just <> itself?
06:24.48phpboyI installed asterisk@home (CentOS)
06:24.58phpboyHad quite a lot of crap with it
06:25.04*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
06:25.05enderphpboy: thats asterisk@home.
06:25.13phpboyLook, I'm willing to learn... I just need a nice howTO :<
06:25.18enderphpboy: isn't there a port of Asterisk for BSD?  FreeBSD I think?
06:25.33enderphpboy: try #gentoo, but brace yourself for the flamage.  THey're not always kind.
06:26.19twistedender, it's a bit more complicated than that ;)
06:26.29phpboyNot many geeks with skill are :<
06:26.36twistedender, but when it's being set manually, it's parsed out of the <>'s
06:26.38enderphpboy: I'm trying to be (:
06:26.39heath__lucasjb: yes, makes sense now
06:26.51endertwisted: hrm, so I can't get away w/ just <>
06:27.04littleballender, just now, you said digium got confused . what do you mean?
06:27.06enderphpboy: honestly, a Linux beginner shouldn't begin w/ Gentoo.
06:27.09heath__i think that without a third party app, the only way to do what you're saying is to make extra Queues
06:27.49twistedender, right
06:27.49enderlittleball: digium thought you were talking about a different blinking light I bet.  Not the identification light.
06:27.49twistedender, feeling up for some code hacking right quick to see if i'm right?
06:27.49lucasjbheath__, OK, what about penalties? Can they be used to acheive something similar?
06:27.49enderphpboy: A distro like Fedora or CentOS would be better off, PLUS they come w/ isdn already in the kernel.
06:28.07endertwisted: well, I no longer have my x86_64 dev env, so I can't build any packages for the production boxen.
06:28.09phpboyeish
06:28.13littleballwhat is the function of identification light? pls see http://supportapj.dell.com/support/edocs/systems/pe2850/en/ug/t1390c10.htm#wp1043338
06:28.17phpboywhere can I get CentOS?
06:28.26enderphpboy: http://www.centos.org
06:28.28heath__by that i mean: make a AFIRST queue that has 'a' in it. a call intended for 'A' will actually check 'AFIRST' first, if it times out, then tries to queue to 'A'
06:28.37heath__penalties... maybe
06:28.44enderphpboy: #centos also exists and they are usually quite friendly and helpful, when there is somebody there.
06:28.46littleballender, i am talking about the "system status indicator" on the back
06:29.00heath__would have to think about that some
06:29.00twistedender, you can't change it on the box that's sending the calls?
06:29.17enderlittleball: ah, ok.  Well, thats most likely an issue to bring up w/ Dell, rather than Digium.
06:29.50endertwisted: same deal, x86_64 production (:  THese are identical systems, one was used to develop / build the packages for both, when they went into production I had to strip the build env out.
06:29.50littleballender, but if i didn't load the asterisk, the light is not blinking
06:29.56phpboyI just reinstalled to Gentoo :<
06:30.04lucasjbheath__, OK, I think I understand your AFIRST idea...
06:30.07enderoh SHIT, I just realized that I f'd  up and didn't even save the SRPMS!  GOD DAMNIT>
06:30.09twistedender, ah.
06:30.11littleballender, ok. what does this mean "The identification buttons on the front and back panels can be used to locate a particular system within a rack. When one of these buttons is pushed, the blue system status indicator on the front and back blinks until one of the buttons is pushed again."
06:30.37jcohenhey If I have a tdm400p with 4 fxo addon boards, how do I set it for dial out if zap/1 is busy to use zap/2 or zap/3?
06:30.39enderlittleball: that means you push the button on the front or the back, and it blinks the light.  Allows you to go to the OTHER side and easily identify which of the systems you were working on.
06:30.59enderCRAP.  That was a good 4~ hours of work I just lost.  *grumble*
06:31.08littleballender, i see
06:31.12redaxgood morning,
06:31.17jcohenphpboy, you could always get support for gentoo from gen-ux (the company I work for)
06:31.28littleballender, only used when i go from the frond to back ....
06:31.54zobiahello everyone . where is the voicemailmain default gsm file stored? i want to use those files. like 'comedian mail, mail box'
06:32.32enderlittleball: pretty much.
06:32.36heath__jcohen: look into zap groups, you can set groups in zapata.conf so you can Dial(zap/g1/231231213)
06:33.06_omerwhat's the difference between ASTWIND and Asteriskwin32
06:33.08jcohenahh
06:33.22enderzobia: /var/lib/asterisk/sounds/vm*
06:33.34littleballender, then it should not a problem for it keeping blinking, right? I see it is some problem there because it keeps blinking once i run asterisk
06:34.12enderlittleball: well, if it is doing more than just a blue blink, it may be an indication of something else, so you should call Dell and ask them.  Thats why you buy dell right, so that you can get the support?
06:34.31twistedender, well, the fix (if it's correct) is simple
06:34.35endertwisted: I hope to have an x86_64 dev box up soon, so that I may test some stuff out.
06:34.58endertwisted: I had to heavily modify some HEAD srpms to build a clean rpm for 1.2.0beta1, so...
06:35.11twistedew... rpms.
06:35.12endertwisted: I'm also on vacation next week, so I may have to wait until after that.
06:35.28endertwisted: heh, shush.
06:35.57_omeranybody:::: what's the difference between ASTWIND and Asteriskwin32   ???????
06:36.03jcohendo I use spanmap or trunkgroup?
06:36.14twistedender, at least do me this favor
06:36.17endertwisted: sure
06:36.23twistedender, open a bug at http://bugs.digium.com
06:36.36twistedi'll go through and put my pointers in, and maybe it'll get fixed correctly ;)
06:36.41twistedsince we can't really test my solution right now
06:36.52littleballender, then don't know. I called them and they just refund me the third external network card and let me to buy an third party network card.
06:36.57endertwisted: sounds good.
06:37.00twistedender, cool.
06:37.01littleballs/then/they
06:37.21twistedender, just give them an overview of what we went through, and the results of the ${CALLINGPRES} test
06:37.25enderlittleball: um, they don't know why the light blinks?
06:37.27jcohenheath__, do I use spanmap or trunkgroup?
06:37.45littleballender, they told me that something wrong. that is all
06:38.28littleballender, actually, this is my first time to try dell server. it is cheaper than IBMs, but i never have such problem with IBMs
06:38.36*** join/#asterisk darkskiez (n=darkskie@host86-132-171-33.range86-132.btcentralplus.com)
06:38.52darkskiezender: the ID light is if you have processor ID enabled, afaik. totally pointless
06:38.53endertwisted: I'm filing against IAX, k?
06:39.00twistedender, perfect
06:39.36darkskiezender: oh, in that one, that lets you light up the LED remotely so if you have 10's of servers + you can verify which one you are accessing
06:39.41*** part/#asterisk redax (n=redax@catv-50621cc0.catv.broadband.hu)
06:40.03littleballdarkskiez, you are talking about the dell system indicator , right?
06:40.20*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
06:40.28darkskiezno, the system identifer light
06:40.58jcohendo I use spanmap or trunkgroup for zap groups??
06:41.02darkskiezblinking blue - thats what it says that means
06:41.12heath__jcohen: i think all you have to do is specify the group right before the channels so like group=1 and then specify all the chans channel => 1 .......  i could be wrong cuz it's been a while
06:41.16darkskiezyou may have a button on the front to stop it
06:41.27heath__i think that spanmap/trunkgroup jazz is for something else
06:41.33darkskiezcrap, gotta go,
06:41.36jcohenahh
06:41.37littleballdarkskiez, do you mean when i access the server, the light will blink?
06:42.07endertwisted: is it a 'bug' that the number is being set to asterisk@ ?
06:42.26twistedender, no
06:42.27jcohenthanks
06:42.39endertwisted: Ok.
06:42.52twistedender, it's a bug that the presentation is getting blocked if there's not a number
06:43.20endertwisted: right, I was thinking secondary bug, as I want no number info at all to show on the phone, just the 'Outside Caller'
06:43.30zobiathank you ender
06:44.32jcohenif 2 rules like:
06:44.48jcohenexten => _1700,1,DIAL(xxx)
06:44.59jcohenexten => _1.,1,DIAL(xxx)
06:45.14jcohenwill _1. execute if the number is _1700xxxx?
06:45.42endertwisted: um, what 'disclaimer' is the bug system asking me about?
06:45.52twistedender, dn't worry about that, select n/a
06:45.55endertwisted: thanks.
06:46.15enderjcohen: seems like it would.
06:46.45jcohenhmm so how do I set it so 1700 numbers route through a special dial method and others run through another?
06:46.47endertwisted: http://bugs.digium.com/view.php?id=5261
06:47.04enderjcohen: put a line above it that catches _1700.
06:47.17twistedender, cool. I'm gonna post a note on it and go to bed.
06:47.31jcohenlike?
06:47.38enderjcohen: your first line will catch 1700 exactly, the _1700. will catch anything else that starts w/ 1700, and _1. will finally catch anything else that starts w/ 1
06:47.55enderexten => _1700,1,Dial
06:47.59enderexten => _1700.,1,Dial
06:48.03enderexten => _1.,1,Dial
06:48.24jcohenright but _1. will execute even if the number is 1700xxx?
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06:48.53enderjcohen: no, because that will be caught by the new line: _1700.,
06:49.05jcohenahhh
06:49.41jcohenok
06:49.43jcohenthanks
06:52.33twistedender, bugnote added, g'nite, and glad to see we were able to at least work around the problem :)
06:52.45endertwisted: yep, thanks VERY much
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07:05.31lucasjbI'm confused about how to add members to groups. agents.conf just says 'Group memberships for agents', but how do I define which members should be in which groups?
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07:09.27szerhi
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07:19.12lehelhello
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07:32.24jcohenwhich codec should I use for my IAXy devices when they are on ADSL lines behind home routers?
07:33.09clive-iaxy can only do one codec g711
07:34.10jcohenreally because the iaxy docs list ulaw and adpcm in the examples
07:35.00dudesYou could use a asterisk box to transcode to g729 if bandwidth is a issue
07:35.19dudesg729 from your provider to ulaw to the iaxy device that is
07:36.09jcohendo you think ulaw would work good in a home adsl 384/384 line behind a linksys/dlink router?
07:36.17drrayadpcm is a 48k version
07:36.25jcohenI have 4 remote home users
07:36.41jcoheneach at a different home, most with adsl and the slowest is 384/384
07:37.34dudesI can get 2 ulaws on a 256 cable no problem
07:37.44*** join/#asterisk NoNeo (n=Artur@193.24.24.10)
07:37.48jcohenthats good to hear
07:38.02dudesNow you can get a lot more /w g729
07:38.06jcohenI am trying to use IAX instead of SIP/VoIP since the lines are async
07:38.30jcohendo you think IAXy can do g729?
07:38.36dudesilbc is good too if it's just asterisk to asterisk
07:38.59dudesI connect my ATA to my asterisk box ... so if you're going to your provider to stuck with ulaw
07:39.16drrayjchohen - you could use openwrt firmware on a linksys router and run asterisk on that, then you get QOS and any codec you want
07:39.42drrayQOS will help more than a lean codec in my opinion
07:40.00dudesHow much space do those linksys's have
07:40.16jcohenthey are not my routers, some have adsl and cable modems, some have linksys routers, some have dlink, etc
07:40.18drrayI want to say 32m on mine
07:40.55FuriousGeorgeu could always put ipcop on an old 200mhz box u got laying around
07:41.14drrayhe has remote locations
07:41.22jcohenthe asterisk box is on a dedicated internet line with over 100MBps
07:41.29FuriousGeorgeah
07:41.38jcohenso its a matter of the remote users
07:42.12drrayjcohen - just try connecting to the server and using the milliwatt app, that will tell you how you are doing bandwidth wise
07:42.14jcohenthey only need 1 line and at most 2
07:42.27jcohenmilliwatt app?
07:42.45jcohena url please
07:43.09drrayexten => 8663,1,Milliwatt
07:44.09jcohenoh milliwat is a module in asterisk
07:44.14jcohenahh
07:44.40drrayif you call that from your remote box you'll get a tone
07:44.48drrayand you can hear if you are losing bits
07:44.58jcohenahh
07:45.23jcohenif I am then I should switch to another codec then
07:45.34jcohenuntil I find one thats decent
07:45.36drrayor get QOS on your remote routers
07:46.16jcohenok now back to the codecs for iaxy, so iaxy supports which ones then? is there a URL with a list of the ones that it supports?
07:46.26drraydigium.com
07:46.33*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
07:47.27drrayit supports ulaw and adpcm (but I've read that the adpcm is 48k not 32k)
07:47.50*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
07:48.45jcohenok, so I should go with ulaw then 1st
07:48.45jdv79shido6, a word when you get a chance
07:49.34jcohenthanks for the advise
07:49.44jcohencya, off to play with the iaxy
07:50.25drraythat boy was our only hope
07:50.38jdv79there is another...
07:54.00*** join/#asterisk _omer (i=o@203.215.180.250)
07:54.51_omerwhen I do transfer() ,,,,, I dont get any ring,,,,but the sound when connected....
07:55.01_omeranyhelp?
07:56.01*** join/#asterisk bmg505 (n=leon@rndf-146-25-51.telkomadsl.co.za)
07:56.59_omerwhy asterisk dont produce the rings to the caller when dial another Asterisk SIP Peer....
07:59.31opus_<bmg505:#asterisk-unregistered> I think u must register your nick before we can talk on it
07:59.37opus_its the bizzaro #asterisk!!!
08:00.07*** join/#asterisk Abbas (n=Abbas@203.81.221.21)
08:01.28bmg505eeeish split channels and combined?
08:02.57bmg505something from th twilight zone :)
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08:03.47clive-aish, bmg:)
08:04.42AbbasSep 21 04:03:37 NOTICE[17263]: frame.c:128 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
08:04.49Abbashow to avoid this message
08:05.55clive-abbas are you using digiums g729 version?
08:06.42Abbasno just testing an open
08:10.13clive-the ipp version?
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08:10.40opus_abbas turn off VAD on your phone
08:11.11t0pHi, how can we identify a call from ZAP channels?
08:11.26Abbashmm  actually i am using Xpro to dial   and my asterisk sending call on Quintum to terminate it
08:11.32opus_set a variable on all incoming zap calls, and check for it
08:11.34t0plike whether it's from ZAP 1 or 2 in the dialplan
08:11.34Abbaswhere should i turn off VAD
08:12.02t0popus_: like caller-id?
08:12.12opus_t0p sort of
08:12.36t0popus_: Ok, will try it out
08:13.03opus_man
08:13.16opus_i just caught  a scammer trying to fuck up my business
08:13.31opus_and now i'm trying to get him to get psyological help
08:13.33opus_hehe
08:13.50FuriousGeorgewhat was the angle
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08:14.09opus_you send $5000, then they send you to "ceo training course" in L.A.
08:14.39opus_then, you will meet other people who spent $5000 at this ceo training school
08:14.54opus_hehe ceo training course. its called IBI Global
08:15.15FuriousGeorgeoriginal, at least
08:15.35Abbashow to turn off VAD on Xpro
08:15.50opus_Abbas your going to have to tell us.. check the manuals
08:15.57opus_Abbas, it shouldn't be that hard
08:16.10opus_look for Voice Supression, Silience Supression
08:16.15opus_suppression arge
08:17.20opus_i think I can convince the scammer to see the light... i have no idea why i am helping him
08:17.35_omerAbbas: turn the silience suppression off
08:18.07_omerAbbas: VAD is off by default in asterisk.....
08:18.19normal1any of you use asterisk on openbsd? or any other bsd system?
08:19.04opus_no. whats the problem though i'm a bsd guy
08:19.10blitzrageevening all
08:19.35opus_hey blitzrage
08:19.36normal1none, I just wanted to see if there were at least some people that did
08:19.58blitzragecan't believe I'm still up programming... :)
08:20.01normal1I'm actually a little hesitant to start asterisk since I Know jack-nothing about phone systems
08:20.04_omercan I produce "rings" in dial() ???
08:20.07opus_normal1 never tried it.
08:20.21normal1opus_: I know one person who runs it under openbsd
08:20.24blitzrage_omer: yes - show application dial() - look for the 'r' flag
08:20.39opus_normal1, curious why that person runs it under openbsd
08:20.51_omerringing() doesnt give the ring tones back to the dialer....ok I check r flag
08:20.59opus_asterisk is full of buffer overflows.. did he think it would be more secure?
08:21.02normal1I havent the slighest clue.. I think security? and probably because they already ran it for their gateway
08:21.17normal1he probably didnt wanna install a new OS
08:21.21opus_omer - you should be careful with "r" because for some reason it screws up a lot of things
08:21.34_omerfor example?
08:21.34opus_omer - like, calls will continue to keep dialing after the user hangs up.
08:21.51blitzrageopus_: I don't tend to get that when I create my dialplans correctly :)
08:21.57opus_and leave an exteremely long voicemail message with nothing in it
08:22.13blitzragegenerally it shouldn't be needed though...
08:22.15opus_blitzrage whats the correct way :)
08:22.34blitzrageopus_: nothing crazy - just handle hang ups with 'h', etc....
08:22.46blitzrageopus_: I never get blank voicemails and hung calls with 'r'
08:22.51opus_hehe yeah, like create infinite loops with h,1,Hangup ? :)
08:22.57blitzrageopus_: exactly
08:22.59blitzrage:D
08:23.25opus_i'm stumped on what it should really do
08:23.41opus_you got a good dialout example with h,1?
08:23.48normal1asterisk requires IP phones to work?
08:24.05opus_normal1 yes, generally speaking.
08:24.16normal1anyone know where I can find a full diagram for asterisk?
08:24.18normal1opus_: ah
08:24.20opus_softphones, ata's, etc
08:24.25opus_can be used as well
08:24.42normal1wonder if one can just use the home phone
08:24.43blitzrageopus_: funny thing is I actually do have a h,1,Hangup() line in my dialplan :)
08:24.47opus_but you want to use IP phones, don't ever touch the analog stuff
08:24.53opus_blitzrage haha
08:24.54normal1ah
08:24.56blitzrageopus_: but thats because I'm calling a Local() channel
08:24.56t0popus_: I have usecallerid=yes,hidecallerid=no,callerid="PSTN-(333)" <333> in the zapata.conf
08:25.07t0popus_: isn't this enough
08:25.19opus_t0p - you want to use dial plan logic
08:25.46opus_t0p - for example, zap context=from-analog-world
08:26.15opus_[from-analog-world]\n s,1,Answer \n s,2,Set(FROMZAP=1) \n
08:26.26opus_you'll need to look up the syntax for Set though thats wrong
08:27.45drraywhat's wrong with the analog stuff?
08:27.50opus_t0p - you could use dial plan logic and regular expressions to look for *PSTN*
08:28.01t0popus_: you mean add different context to zapata.conf for each ZAP channel?
08:28.34opus_t0p - just one, and then Goto other ones based on combination of If statements ?
08:28.56opus_drray - its a bad investment
08:29.29drraywe'll be using zap stuff 25 years from now
08:30.16normal1is there an asterisk how to some place
08:30.25drrayvoip-info.org
08:30.39normal1tx
08:32.24opus_we'll be using zap stuff 25 years from now, but it will all be a hack
08:34.30johnmanyone know of a text 2 wav wrapper for linux?
08:34.49wasimjohnm: you mean a text to speech program?
08:35.16opus_festival.
08:35.36johnmwasim: I actually meant a text 2 wav wrapper, which will end up using at least one txt2speech app im sure.
08:36.16johnmactually.. Asterisk has a festival module doesn't it? :)
08:36.43opus_can anyone recommend a good C programming book. i've read all the classics, i just need to roughen up a bit
08:37.04opus_like, why in the hell are there periods "." before variable names in structures nowdays
08:37.26*** join/#asterisk cjk (n=cjk@80.92.64.103)
08:38.03cjkhi, is there a way to modify the context which is written into the cdr? for example i can define the userfield, the accountcode, is there no way to change the context
08:38.57johnmopus_: does * have a way to define a variable with text from a file?
08:39.44opus_johnm a dial plan variable?
08:40.41johnmopus_: well, anything. Maybe just a way for Festival to read the contents of a file, rather than a string.
08:41.12opus_Festival by nature reads contents from a file
08:42.24johnmopus_: not within the dialplan
08:42.47opus_There is a AGI script that calls festival which you can modify
08:43.02opus_agi scripts can read , and I just learend this, and write dial plan variables
08:43.17opus_festival sounds like crap though
08:43.40FuriousGeorge.
08:44.07opus_ahhh! bizzaro asterisk channel
08:44.10drraywe'll be stuck with zap at work for the foreseeable future because it will cost too much to rewire the building
08:44.48*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:45.39normal1"You do not need need any special hardware to install and run Asterisk. "
08:45.52normal1I thought you need it some sort of FXO card or somesuch
08:46.04drrayno, you can sip/iax without it
08:46.15jdv79for voip - think real world dude
08:46.17drrayyou need ztdummy
08:46.29drrayor a fxo card for timing
08:46.55normal1either or will work?
08:47.07drrayI've never tried ztdummy
08:47.17drraybut I know the x100p card works for it
08:47.26normal1http://www.digium.com/index.php?menu=product_detail&category=hardware&product=DEVKIT
08:47.28normal1whats that for?
08:47.37opus_i bought a x100p clone for $3 on ebay and it worked, then i just dumped it for ztdummy
08:48.08jdv79normal1, you're serious?
08:48.11drraynormal - that gives you a FXO port (to the PSTN network) and an FXS port (a plain extension)
08:48.22normal1thanks drray
08:48.37drrayyou can bypass the FXO port if you get a sip provider (or an iax2 provider)
08:48.38normal1jdv79: Why wouldnt I be?
08:49.02normal1I'd like to eventually get an voip line and use asterisk as a gateway to it
08:49.03drrayjdv79 - it this the "another" you spoke of? :)
08:49.05jdv79i don't know, maybe cause the page says what it is
08:49.12pauldyany of you guys know of a cheap fxs pci card
08:49.18jdv79if so we're all doomed;)
08:49.19pauldyseems almost cheaper to just but an ata
08:49.36tengulreLONG LIVE THE KING!!:)
08:50.04pauldybut/buy its gettign late
08:50.37drraypauldy - you have to do the math on that
08:50.45drrayit was cheapest for us to buy a channel bank
08:51.01pauldydrray, just looking for a single port fxs interface
08:51.21drrayI've been happy with the IAxy
08:51.27jdv79channel banks are 1K+ aren't they?  like carrier access?
08:51.29pauldypreffreably pci but the ATAs seem to be cheaper than cards like the TDM410b
08:51.50pauldywhick appears to be the cheapest fxs pci card i could find
08:52.01drrayjdv - we are using zhone zplex's for $120 each
08:52.16pauldydrray how many phone are connected to your iaxy
08:52.25drrayone per Iaxy
08:52.38jdv79oh, i'm talkin old skool telco style then...
08:52.51pauldyyea I'm looking at the grandstream 386
08:52.51drraythe zplex's are junk
08:52.56drraybut they work
08:53.13drrayI want to get work to upgrade to an Adi 600
08:53.14jdv79i've only ever used adtrans or carrier access
08:53.16drrayerm Adit
08:53.17pauldyplan on hooking up at least 4 analog phones to it
08:53.55drrayare you faxing/modeming?
08:54.12pauldydrray, me no
08:54.20drraywhy not just buy IP phones?
08:54.26drrayif you don't have a distance problem
08:54.40pauldywant to use existing cordless phones
08:54.48drraygood enough
08:55.06drrayyou'll have some issues with nat traversal if you go with a SIP ata
08:55.25pauldywhy if it is on my internal network connecting to asterisk?
08:55.42drraythen that might be a good plan
08:55.51drraythey sell ata's that have 2 fxs ports on them
08:55.59drrayI've never used them though
08:56.23pauldyyea I saw that but I don't think given the education i got on REN last night that will be much of a problem
08:56.35pauldybut someone did mention not to ever do this with an iaxy
08:56.53drraywell, you'll just need 4 iaxy's
08:56.56drray:)
08:58.08pauldyI would have just preffered it stay in the asterisk box
08:58.20pauldybut for the price it looks like the ata is the way to go
08:58.57drraywell, the good news is you wont outgrow an ATA
08:59.11drraywe outgrew our TDM40b
08:59.40pauldytrue, hows the latency though ever have a problem with that
08:59.55drrayI don't
08:59.56opus_don't use analog
08:59.59drraybut I use QOS
09:00.14opus_don't use analog --> analog == bad <--
09:00.39pauldyopus_, ?
09:00.46drraywe have a cisco 7960 which is plugged into Asterisk via a crossover cable..
09:01.07opus_pauldy - analog is bad
09:01.19drraybut I'm driving 4 payphones with Iaxy's and they work fine
09:01.21jdv79seen shido6?
09:01.27jdv79where's the bot?
09:01.28opus_drray cool
09:02.17pauldywhere at like a bus stop or something
09:02.21pauldyor hotel
09:02.37*** join/#asterisk Umaro (n=umaro@209.140.74.64)
09:02.39drrayhotel
09:02.55Umarohey guys
09:03.05*** join/#asterisk Laerte (n=io@195.47.232.200)
09:03.07Laertehy
09:03.47Umarohaving some issues getting my dialplan setup for these guys to call italy.. anyone have experience that can help me?
09:03.48*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
09:05.03areskihi guys, is there anybody here that knows about h.263
09:06.15areskiI am trying to play h263 files with asterisk but dont succeed except if there are previously recorded with voicemail.
09:06.31*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
09:06.45areskiI guess it s something to see with the payload header...
09:06.58areskiany advice on this ?
09:11.12*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:16.23NoNeoHi all. I'm looking for expalanation, how remove latency of voice (1-5secs) in linux IAX client. It appeares regardless of connection type Fast Ethernet or ADSL 64kbps.
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09:32.30drrayNoNeo are you hooked to a zap channel at all?
09:35.35NoNeoI'm not familiar with the hardware (my friend set it up), but yes - something like zap(ata) is configured and works well. It is all right with windows client, the problem is on Linux. May I change the latency in asterisk using a "hidden" option? Changing TOS in iax.conf does not work.
09:39.42NoNeoI have testing configuration: hardware telephone -> Avaya E1 -> Asterisk E1 -> IAX client, or IAX client -> Asterisk -> IAX client. Both enviroments couse the problem when I use Linux client.
09:40.45*** join/#asterisk RoyK (n=roy@80.239.107.80)
09:43.14olivier_Noneo, under Linux, when several app try to access the sound card, it make very buggy sound.
09:53.49NoNeoYes, it is an idea. I think about recompiling kernel. May be I'll find some option about real time access to devices. People working on sound applications need such conditions, so I hope I'll find some tips on the Net.
09:54.07*** join/#asterisk djin_ib (n=djin_ib@84-245-25-231.dsl.cambrium.nl)
09:55.29NoNeoolivier_, but changing an option on asterisk is more pretty :-)
10:02.40NoNeoolivier_, I think about your point, and I remember, that swiching between application make sound distortions, but it is not latency/delay.
10:11.06*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
10:11.22puzzledmorning
10:11.30RoyKmorning
10:13.51*** join/#asterisk key2 (n=tree@gob75-2-81-56-64-17.fbx.proxad.net)
10:23.55*** join/#asterisk sozo (n=vidar@pppoecl75075.minlos.no)
10:23.56sozois there any way to get bristuff to work with CVS HEAD?
10:24.36sozo(beside going through about 100 failed patches
10:30.06puzzledsozo: iirc there are some updated patches. search the mailinglists
10:30.36sozook. thanks
10:33.13*** join/#asterisk vlrk (n=vlrk@59.93.69.196)
10:33.47vlrkdo we have any other option to relaoding the modules other that reload <so file name of module name>
10:39.22RoyK'restart now'
10:39.24RoyK:P
10:40.13vlrkok
10:40.19vlrki will try
10:40.43vlrkyou mean that asteirsk -rx"restart now app_voicemail.so" like that
10:40.54RoyKer
10:40.57RoyKno
10:42.01vlrkso ?
10:42.36vlrkwe donot have any other option to re loading  the modules except "reload"
10:53.23*** join/#asterisk fabrizioxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
10:53.45fabrizioxxxhello all.. can i unregister a sip peer from the asterisk CLI?
10:55.55johnmdoes anyone know if callfiles can dial a macro?
10:56.04johnm(instead of a Channel)
10:56.26djin_ib/usr/src/linux-2.4/include/asm/system.h: In function `__set_64bit_var':
10:56.26djin_ib/usr/src/linux-2.4/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules
10:56.39djin_ibDid anyone see that before (while compiling zaptel)?
10:56.52djin_ibon debian 3.1 (2.4 kernel)
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11:03.20*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
11:10.33RoyKdjin_ib: what arch?
11:10.47RoyKfabrizioxxx: don't think so, no
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11:19.33*** join/#asterisk langals (n=icechat5@196.7.14.183)
11:19.59langalsHi there....wondering if anyone has got speex working with asterisk on ubuntu linux?
11:23.59*** join/#asterisk morlac (i=morlac@213.186.161.28)
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11:25.35joelsolankiHi all, need suggestion/help any weblink ..i need to setup voip billing system in asterisk wheather in mysql or postgresql
11:25.44joelsolankican anybody share things??? plz
11:25.54morlacany idea why MFC/R2 ftp been down? any other recent mirrors around?
11:26.29joelsolankiAny ideas ?
11:27.29Ahrimanesjoelsolanki: prepaid?
11:27.34joelsolankiyes
11:28.03Ahrimanesjoelsolanki: http://www.voip-info.org/tiki-index.php?page=ASTCC <- i'm using this at the moment, quite easy to setup
11:28.52morlacany1 know how I can add my experience with heavy calls on Asterisk/Zapate to Wiki on Voip-info?
11:29.26morlacam talking about 120 Pri calls on E1 hitting Digium hw in less than 60 seconds
11:29.28joelsolankiahrimanes: i tried http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2 but it is little confusing.
11:30.06Ahrimanesjoelsolanki: yes i also tried it, astcc was simpler
11:30.21wasimmorlac: add it to the large-asterisk page
11:30.46morlacwasim: how? need to register or somehting?
11:30.50joelsolankiAhrimanes: so u r running astcc ? working good ?
11:30.54wasimmorlac: yep, you need to register first
11:31.13Ahrimanesjoelsolanki: working fine yes
11:31.16morlacwasim: thanks man....of to writing my exp.
11:31.19morlac:)
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11:33.51X-RobI can be such a prick sometimes.
11:34.02joelsolankiAhrimanes: have u implemented and changed any code accordingly to your requirement or it is same code.
11:34.23X-Robmorlac - I'm guessing your 120 calls in 60 seconds doesn't work?
11:34.27Ahrimanesjoelsolanki: small changes
11:34.43Ahrimanesjoelsolanki: changed it to postgresql instead of mysql.. but that's easy
11:35.07morlacX-Rob> actualy, it did, runnging AGI with perl as well
11:35.11joelsolankiAhriamanes: oh ok what do u suggest for database mysql or postgresql? which is better and stable?
11:35.28morlacX-Rob> the limit is 6 setup calls per second
11:35.36X-Robmorlac - hardware?
11:35.45Ahrimanesjoelsolanki: i put my money on postgresql, but mysql is rather mature too
11:36.15morlacX-Rob> 2GB ram, Xeon 3.4Ghz 1MB cache on Intel SE7520BD2 with SCSI mirror in software
11:36.29morlacX-Rob> 2nd Gen TE410p
11:36.35X-Robjoelsolanki - MySQL is better for smaller single-instance applications. Postgresql is more an industrial database. Not quite as fast, but much more robust when you're doing lots of things at the same time.
11:36.46joelsolankiAhrimanes: well i have prepaid system in mysql using with gnugk. but i dont find it much stable. database size is only 100 mb but still mysql uses 80% of cpu
11:36.50X-Robmorlac - T1 or E1?
11:36.59morlacX-Rob> 4xE1
11:37.22morlacX-Rob> the E1 are coming from antoher Asterisk machine with same specs.
11:37.54morlacX-Rob> is that good?
11:38.08X-RobThat's pretty good, I'd guess. I'm trying to think of the packet size of a call setup.
11:38.13X-RobI think it's about 2-3k.
11:38.26X-Robnah
11:38.31morlacX-Rob> res_perl crashed the whole system running that load
11:38.32X-Robit's not that big, it's about 700 bytes
11:39.03morlacX-Rob> the AGI scripts query Postgres database when it runs
11:39.38X-Robmorlac - I'd be doing load balancing at that sort of level. If you're expecting to be having a whole pile of calls coming in at the same time, I'd be splitting them among machines.
11:39.40Ahrimanesjoelsolanki: oh ok.. can also be the gnugk application
11:39.45X-Robhave a cheap P4 with a TE110
11:39.47morlacX-Rob> I believe an option on Zaptel to tell the system something like CallsPerInterval will help loadbalance in this setuation
11:40.04X-Robtrunk IAX to the big xeon
11:40.25morlacX-Rob> thats out of the question....my limit is only one machine
11:40.57X-Robwhat happens _after_ 6 calls/sec?
11:41.09X-Robasterisk just ignores the call and the caller gets a long delay then a busy, I'd guess.
11:41.42morlacX-Rob> it keeps creating 6 calls per sec with a total of 120 calls. all calls complete with duration of 213 seconds (thats the serivce it self)
11:42.04X-RobAaaah.
11:42.05*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
11:42.27*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
11:42.39X-RobIt might be worth doing a pri intense debug on all the spans and submitting them to mantis (bugs.digium.com) as a 'can't accept more than 6 calls per second' bug
11:44.09morlacX-Rob> believe me when I tell you that it is clear Asterisk/zaptel was not desinged to take such a beating.....all sorts of bugs and errors......for ex. the cables are 2 feet long and I receive HDLC abort errors, spans going offline and reloading....linked list errors..etc
11:45.07X-Robmorlac - no, it wasn't. I totally agree. But it's a feature people want. So submit it as a bug
11:45.23X-Robit will _never_ work unless people actually say 'oi, this doesn't work' where the developers are looking.
11:46.12morlacX-Rob> I will, am writting the report, but that will take some time for now....I also have a solution idea for the short term but I know nothing of PRI-ISDN and zaptel..:(
11:46.30X-Robthere's no 'report'
11:46.32X-Robdo debugging
11:46.35X-Robcompress the files
11:46.42X-Robwrite 2 lines on what you've been doing.
11:46.47X-Robadd to mantis.
11:46.52morlacthats it?
11:46.56X-Robyes
11:47.02X-Robif people want more information, they'll ask for it.
11:47.14morlacgreat, saved a lot time you did ;)
11:47.38morlacan eay way to dump the pri debug to external file?
11:48.01X-Robedit /etc/asterisk/logger.conf
11:48.09X-Robfull => notice,warning,error,debug,verbose
11:48.14X-Robthat's your logfile
11:48.27morlacthen /etc/log/asterisk/messages is the file?
11:48.37X-Robusually /var/log/asterisk/full
11:48.38*** join/#asterisk NoNeo (n=Artur@193.24.24.10)
11:48.48morlacX-Rob> thanks man
11:48.59X-Roblook at 'astlogdir' in /etc/asterisk/asterisk.conf
11:50.36morlacX-Rob> btw, if I dont use AGI and dump the 120 calls in one second, there is 99% chance that the whole calls will complete
11:51.15morlacX-Rob> but sometimes it fails all/most of them
11:51.32morlacX-Rob> anyway, ill do the mantis with the logs
11:51.36X-RobYup.
11:51.43X-RobIt's not going to get fixed if you don't report it
11:51.49X-Robbut I think you're running into hardware limits
11:52.17X-Robor, possibly, some slight deadlock in spawning an AGI thread or something.
11:52.24X-Robbut make sure you mention that in the report!
11:53.10*** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
11:53.57*** join/#asterisk valence_ (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
11:55.16morlacI will include everything and all details...
11:55.35*** join/#asterisk xming (n=xming@gentoo/user/xming)
11:55.58*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
11:56.16morlacjust to let you know, this was impossilbe on a dual xeon 3.06Ghz runing kernel 2.4.29 with lowlatency patch
12:04.09djin_ibRoyK, very late response, zaptel v1.0.9.2
12:04.35RoyKno
12:04.45RoyKwhat cpu architecture?
12:04.57RoyKamd64? ia32?
12:05.12djin_ibOh sorry :) Dual Xeon, x86
12:05.20djin_ibno 64bit kernel.
12:05.21RoyKxeon32 or xeon64?
12:05.23RoyKok
12:05.28RoyKwhy not????
12:05.35RoyKif you're running xeon64, that is
12:05.50RoyKintel-branded amd64
12:05.51RoyK:P
12:05.53djin_ibThey are 64's, but I need to run a Eicon board on it as well.
12:06.41djin_ibI have no experience with 64, but I assumed there was no/little advantage.
12:08.54*** join/#asterisk Jas_Williams (n=Jason@86.130.0.82)
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12:11.13drrayok, tiddlywiki is kickass
12:15.02*** join/#asterisk szer (n=szer@217.116.36.22)
12:15.16*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
12:15.29lathos42Morning
12:15.33RoyKdjin_ib: there's LOTS when it comes to transcoding
12:15.48RoyKdjin_ib: also, much cleaner memory addressing > 1GB
12:15.49*** join/#asterisk lehel (n=asd@82.79.20.17)
12:15.52lehelhello
12:17.01festrhello
12:17.19festranyone have checked this? : http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre1/
12:17.26festrt38-bits.tgz
12:17.59festrvery interesting
12:18.03festrbut no documentation
12:18.04Ahrimanesooh
12:18.27*** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
12:19.36lehelfestr: what do you mean interesting?
12:19.59Ahrimanest.38 support would be VERY interesting
12:20.13festrlehel: interesting, that there is some test application for udpt and no information provided anywhere
12:20.35lehelt.38 is a kinda virtual fax thing with asterisk? or what?
12:20.41*** join/#asterisk ovanoide (n=RobertoO@host205-72.pool8174.interbusiness.it)
12:20.58festrlehel: yes
12:21.16festrlehel: t38 is only one way to support faxing over voip
12:21.17*** join/#asterisk Laerte (n=io@195.47.232.200)
12:21.23festrlehel: but it is not free
12:22.08Ahrimanespstn <-> asterisk <-> IAX2 <-> asterisk <- should the last asterisk here be able to receive fax with spandsp?
12:22.14djin_ibRoyK, ok. But how about drivers? Like the Eicon -> compiled from src.
12:22.42RoyKlehel: see the wiki
12:22.49RoyKfestr: t.37 is nice too
12:22.52X-RobAhrimanes - No.
12:22.56X-RobVoIP and fax does not work.
12:23.05festrX-Rob: with t.38 yes
12:23.05X-Robsee all those '<->' where the '<->' is IP?
12:23.08*** join/#asterisk magglz (n=michi@p549652A1.dip.t-dialin.net)
12:23.12RoyKfestr: or t.37......
12:23.16festror
12:23.17RoyKt.37 is really a lot better
12:23.24festri dont know what t.37 is
12:23.30RoyKstore-and-forward instead of realtime
12:23.30festrwhat is differ?
12:23.30X-RobIt may work. Possibly. But the correct answer is 'no'
12:23.55djin_ibthat
12:24.08festrRoyK: so it is received after called to other side?
12:24.26Ahrimaneshey djin_ib :)
12:25.46lehelX-Rob: do you mean that PSTN -> asterisk (chan_capi/ISDN)-> Zap -> Fax ..isn't works?
12:26.04X-Roblehel - where's the IP there?
12:27.31lehelX-Rob: sorry, that do you mean where? in wich country .. ??
12:27.37lehel*what
12:27.44X-Roblehel - you offered something saying 'this won't work'
12:27.49X-RobI can't see where the Internet Protocol is
12:27.50X-RobIP
12:28.52lehelbetween the PSTN and asterisk-box (?)
12:28.58X-RobYou said that's ISDN
12:30.49lehelasterisk-box has the IP
12:31.04X-Rob<PROTECTED>
12:31.10X-RobThere is no Internet Protocol connection there.
12:31.44X-Robif you unplugged the network cable from the machine, it would still work
12:31.48X-Robthere's no VoIP happening.
12:33.35djin_ibHi Ahrimanes
12:33.37djin_ib:)
12:33.55Ahrimanesdjin_ib: whats up?
12:34.14djin_ibMore, waht's down :)
12:34.21synthetiqSep 21 07:29:52 ERROR[4623]: chan_sip.c:10327 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 12403758223 in context brewster
12:34.28synthetiqwhat does that mean
12:34.30Ahrimanesdjin_ib: hehe whats down?
12:34.53djin_ibsome zaptel compilation errors on Debian
12:34.58X-Robsynthetiq - find [brewster]
12:35.12X-Robadd 'exten => 12304758223,hint,SIP/whatever'
12:35.15*** join/#asterisk Proteque (n=gjorans@213.184.199.245)
12:35.26X-Robor ZAP/whatever, or IAX/whatever. Whatever that device is.
12:35.29*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:35.37Protequehey. how can I see if a iax-trunk is connected?
12:36.01Protequethe nice women keeps telling me that all surcits are buzy :)
12:36.37bjohnsonincrease the logging and watch the cli
12:36.44bjohnsonor try the show channels command
12:36.59bjohnsonor iax peers
12:37.03bjohnsonor iax clients
12:37.11bjohnson(deoending on how it is defined)
12:37.38*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
12:37.58synthetiqxrob: how critical is adding that inr equires
12:38.00synthetiqreqired
12:38.04Proteque0 active channels. strange. I am using the @home-distribution btw.
12:43.19X-Robsynthetiq - you're trying to see if a phone is in use.
12:43.29X-Robif you don't put a hint in, asterisk doesn't know where to look.
12:44.08synthetiqits askign for hints on phones outside the dial plan
12:44.24X-Robtell your phone not to then.
12:44.30synthetiq(im usign cvs head now, before used older versionw ith dial plan and did not have this problem)
12:44.55X-Robhints have changed a lot.
12:45.02X-Robthey now actually work 8)
12:45.06X-Robwhat sort of phone are you using?
12:45.18synthetiqwe are having a current problem with people recv calls on a polycom but hearing nothing,  i am trying to determine if its a firewall issue or this error i keep seeing
12:45.25synthetiqpolycom 501
12:45.31X-RobDunno 'bout the polycoms.
12:45.35X-Robbut it's definately not that.
12:45.39X-Robit's a firewall issue.
12:45.42synthetiqok
12:45.57synthetiqas long as its not my fault, im fine, lol
12:46.14X-Robtry putting 'nat=yes' in their sip.conf entry
12:46.14memiche how can i display certain stuff in the display of my sip phones?
12:46.42*** join/#asterisk buzzyd (n=buzzyd@82-35-251-70.cable.ubr01.enfi.blueyonder.co.uk)
12:46.51memici have 2 diffrent numbers reachable from outside, i want to see which nummer was called on my display
12:47.21synthetiqwell we set a route so nat is not an issue behind our firewall
12:47.22X-Robmemic - depends on the phone. Usually the phone will display the caller ID on the call received.
12:47.23mog_homeone way, is you could add to the callerid name field
12:47.33synthetiqerr firewall i mean asterisk server
12:47.34mog_homesetcalleridname or something like that
12:47.47X-Robso try setting it to... Set(CALLERID(name)=Line1)
12:48.01memiccan i set the caller id in asterisk when i call the phone?
12:48.04X-Robmog_home - depreciated. CALLERID is the lame arse duck now.
12:48.09Protequeasterisk@home makes the sipaccounts in additional files. but in the sip.conf it has a hash infron of the include. is hash like other unixprograms a way to comment the line?
12:48.11buzzydDoes anyone know if it is possbile to have a custom menu in meetme so that the person creating the conference can mute users or switch users to monitor mode all via their handset?
12:48.22X-RobProteque - no. it's the same as in C.
12:48.30X-Rob#include ... means "include ..."
12:48.30Protequeokay
12:48.32Protequethank yo
12:48.35X-Roba comment starts with ;
12:48.59memicX-Rob where to set exacly?
12:49.14X-Robmemic - in the context that the call is coming in.
12:49.16synthetiqx-rob, what is hinting exactly?
12:49.30X-Robif you don't know what that means, you need to do a lot more reading. Spend a couple of days on voip-info.org
12:49.31memici have a extension like this exten => 123456,1,Dial(SIP/phone2&SIP/phone1)
12:49.50memicboth are ringing but and are displaying the number from where was called
12:49.56X-Robsynthetiq - the light on your phone that shows when another phone is in use.
12:50.15synthetiqo cool
12:50.23synthetiqso i dont have to install app_devstate
12:50.38synthetiqto hint other extensions are in use on the snom 360s
12:51.02X-Robmemic - yes, that's right. So, um, geez. I'm not sure. Something like configure your zapata.conf to put each line into a different context, and in that context set the callerid?
12:51.08*** join/#asterisk _omer (i=o@203.215.180.250)
12:51.21memichm ok thanx will have a try
12:52.14*** join/#asterisk oej (n=Olle@64.251.117.114)
12:52.16X-Robbedtime for me almost.
12:52.20X-Robevening oej
12:52.35X-Robseen http://aussievoip.com.au/gxp.jpg?
12:52.42oejMorning
12:52.44*** join/#asterisk rob314 (n=rob314@207.58.194.2)
12:52.44X-Robnew subscribe support in the GXP2000's
12:52.55oejCool
12:53.16synthetiqnow that hinting is working, would picking up a call on that hinting light work
12:53.43X-Robsynthetiq - not just yet.
12:53.45buzzydAny conferencing guru's here?
12:53.54X-Robyou can use app_pickup for that, which was put into CVS just recently.
12:53.59X-Robbut you still have to dial a number.
12:54.11synthetiqi got cvs from yesterday
12:54.34X-Robtype 'show application pickup'
12:54.53synthetiqyup i have it
12:58.08synthetiqalso one more questions, how can sip debuging be turned off
12:58.23X-Robsip no debug
12:59.09synthetiqok thanks
12:59.30synthetiqstupid check poitn firewall
12:59.41synthetiqwe are waiting for the guy with access to it to respond
13:00.05X-Robhave you tried nat=yes?
13:01.08memiceh X-Rob exten => 123456,1,SetCallerID(call for blabla)
13:01.12memichas done the job
13:01.30memicthx again
13:01.54synthetiq<PROTECTED>
13:02.09synthetiqnat=yes kicked the phones off the server
13:05.09synthetiqwhen we upgraded to cvshead yesterday, the check point blocked the phones from registering because there must be new packets sent out by asterisk
13:05.10X-Robsynthetiq - well. Good luck.
13:05.41synthetiqso we can get long distance and lcoal working, just internal does not work
13:06.08synthetiqso we missed some internal setting
13:06.13synthetiqi hate IDS firewalls
13:07.04memicanybody ?has example config for answering machine?
13:07.21memicor howto
13:08.17johnmsynthetiq: if you're talking SIP then there shouldn't be much changed :\
13:08.46buzzydmemic, see www.tomshardware.com they have a simple guide to it
13:08.56synthetiqi dunno we kept or config files the same and jsut upgraded, adn firewall instantly blocked us
13:09.16_omerhttp://pastebin.ca/23510 <---------------any help???
13:09.53memic<PROTECTED>
13:10.08*** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg)
13:10.26johnmsynthetiq: if it's dailing due to an IDS rule, then it should tell you what rule it was which denied it.
13:10.43johnmsynthetiq: if it's not IDS, and it's actually SPI you mean, then it shouldn't be anyu different.
13:10.49johnmsynthetiq: weird though.
13:12.07synthetiqyea well..i dont have access to the firewall...we pay a guy 300$/hr to look at it
13:12.20synthetiqeven though its not our firewall heh
13:13.45littleballsynthetip, any IRC channel or forum for airtime exchange? I am looking for call terminiation service
13:14.39*** join/#asterisk jontow (i=jontow@ws.woflsys.net)
13:15.55_omerlittleball: www.buysellminutes.com ,
13:16.14littleballthanks
13:16.29_omerI had a good website for calls termination but lost the address :(
13:28.35*** join/#asterisk litage (n=nick@203.201.97.87)
13:29.36*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:30.47Ariel_Good morning all
13:30.48*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:31.25znoGdoes anyone have a PAP2-NA?
13:33.44chiardonHello
13:33.53*** join/#asterisk mmlj4 (n=looseduk@ip70-171-92-106.no.no.cox.net)
13:34.06chiardonAriel  . .what about the hurricane?
13:34.32Ariel_chiardon, no problem just a little rain and wind. All is well thanks
13:34.52chiardonbut Rita was declared category III
13:34.56*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:35.18Ariel_It's actually a Cat 4 now in the gulf.
13:35.41*** join/#asterisk iCEBrkr (i=icebrkr@24.129.130.158)
13:35.41Moc_lol...
13:35.54Moc_I am freaking happy to live north..
13:36.02Moc_nothing happen over here
13:36.06NivexI'd like to go back to the north.
13:36.12Ariel_Moc_, yes it does
13:36.28*** join/#asterisk spackle (n=spackle@209.234.83.19)
13:36.33Ariel_have you forgotten about the northeastern storms.
13:38.38*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
13:38.54Moc_Ariel_, those are fun storm...
13:39.05iCEBrkrThis whole getting dumped into a 'unregistered' channel bullshit is fucking gay
13:39.11Moc_I mean just a week without electricity aint the end of the world
13:39.11NivexNortheastern storms just dump a bunch of snow, which eventually melts.  Not a whole lot of damage (save the occasional power line snapping)
13:39.22*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
13:39.26Moc_ice skating everywhere
13:39.37iCEBrkrLilo in his infinite wisdom.
13:39.38spackleMoc_ don't you go into withdrawal from no computer?
13:39.40Ariel_Really. I got stuck in Buffalo NY for 2 weeks snowed in in 1980 please no more of that it's worst then what we get.
13:39.49Moc_spackle, I went to the server room ;)
13:40.07spackleMoc_ for the air conditioning?
13:40.15shmaltzwhy do banks, shop rites, wal marts, and call centers compete for the same stupid ppl? why cant at least one of them hire ppl with an IRQ higher than 60
13:40.16shmaltz?
13:40.28spackleIRQ?
13:40.31Moc_IRQ lol
13:40.34*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:41.00Moc_damn I love this new project... pumping 263mbits/sec
13:41.27Ahrimanesdoing what?
13:43.35Moc_audio streaming
13:43.54znoGdoes anyone have a PAP2-NA?
13:43.56Ahrimanesok, gotta cost a bit
13:44.20Moc_not that bad... but client pay so
13:44.57Ariel_znoG, ask the question are you having issues with it?
13:45.29littleballhi, for manager API, is it possible to send a few Actions in one time and waiting for responses asynchronily?
13:46.09*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:46.09*** mode/#asterisk [+o anthm] by ChanServ
13:46.15Moc_hi anthm
13:46.33anthmhi
13:47.51synthetiqanyone have supura 1001 specs
13:48.23shmaltzsynthetiq, what you lookin for?
13:49.05synthetiqare theys uppsoed to only come up 10meg half duplex
13:51.39Moc_hi mark
13:51.58mutilatoranyone in here play hl2 - cs?
13:52.30synthetiqwhat do u wanna play it over asterisk
13:52.31synthetiqlol
13:52.42mutilatorno
13:52.53mutilatori got a server up and i wanted some people to hop on and test it
13:52.57mutilatorlatency on it and whatnot
13:53.07*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
13:53.13*** join/#asterisk zoid99 (n=cch123@border0hsv.asterisksgi.com)
13:53.40*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
13:54.11EgonisI keep getting 'rejected connect attempt from ip.ip.ip.ip, request 'PhoneNum@incoming' does not exist.. although it does, and yes I did a reload
13:54.29mutilatorhow about ya paste the actual line?
13:54.32zoid99any idea why Voicemail would all of the quit sending to 'a' when '*' is pressed?
13:56.12areskihi there
13:56.45Egonismutilator: socket_read: Rejected connect attempt from 209.91.145.154, request '6477226941@incoming' does not exist
13:57.07*** join/#asterisk bendy24 (n=slb@50.tender.1meg.golden.net)
13:57.37areskiis there anybody here that knows about h.263
13:57.45tzangerwerd to the kram
13:57.49areskiI am trying to play h263 files with asterisk but dont succeed except if there are previously recorded with voicemail.
13:58.00areskiI guess it s something to see with the payload header...
13:58.11areskiany advice on this ?
13:58.19Ahrimanesareski: hm, you recorded with some other program?
13:58.39Ahrimanesareski: same guy i met in madrid? hehe
13:58.50areskiyes probably
13:58.51*** join/#asterisk valence (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
13:58.53areski:D
13:59.10areskiAhrimanes, trying to convert them with ffmpeg
13:59.11znoGAriel_: no, i actually wanted to ask what sort of features the device has (to compare it to a sipura).. like, does it have a dial plan, does it support NAT (on SIP), etc...
13:59.51areskiAhrimanes, what ur name btw ?
14:00.00Ahrimanesareski: michael
14:00.22Ariel_znoG, it's almost the same as the sipura 2000. But the new sipura 2002 is a better unit as it allows 2 g729 use.
14:00.52Ahrimanesareski: hmm.. havent tried ffmpeg for converting yet.. could you send me one of those h263 files that you have?
14:01.04Ahrimanesareski: we've patched our asterisk for better videohandling
14:01.26areskiAhrimanes, ohhh really
14:02.04areskiAhrimanes, are you able to handle converted h263 file ?
14:02.21EgonisNow it says 'Cannot find extension contect 'default' although my extensions.conf is unchanged
14:02.30Ahrimanesareski: not sure, that's why i would like you to send one :D
14:02.45areskiAhrimanes, :D what ur mail ?
14:03.00newmemberanyone use maediatrix 1500 units?
14:03.29Ahrimanesareski: priv..
14:03.39znoGAriel_: yea, that's not really a concern for me as they don't use G729.
14:04.09znoGAriel_: i'll leave them using alaw to the PAP2-NA, and GSM outbound.
14:04.51Hmmhesaysomg I think i should not be at work
14:05.05bendy24anyone know why i have one-way audio on my cisco ip phone?
14:05.06Egonisasterisk is not loading my extensions.conf at all! :O
14:05.06Ahrimanesme too
14:05.22*** join/#asterisk klictel (n=klictel@207.107.208.140)
14:05.32klictelgood morning all
14:05.46Hmmhesaysi'm going to puke
14:05.54Hmmhesaysgood morning klictel
14:06.11spacklebetter get a bucket
14:06.24klicteltoo much stress
14:07.23znoGAriel_: ok, do you remember if you can do dialplan tricks like when you dial a certain number, it changes the sound of the tone?
14:08.30Ariel_your can dial with a r2 option use the show application dial
14:10.56*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
14:11.37znoGAriel_: r2 is to indicate ringing, isn't it?
14:12.21*** join/#asterisk Stephnie (i=Stephnie@u15173939.onlinehome-server.com)
14:13.03tzangerwow
14:13.11Stephnieanyone ?  http://pastebin.ca/23517
14:13.13tzangerI managed ot get contacted by Bell Canada's 911 administration office
14:13.35tzangerthis is the exact # to call when you want to set 911 addresses
14:13.47Ariel_znoG, r2 is what we 2 quick rings instead of normal ring here
14:13.54tzangerof course she's confused because I said I want to tell her which address matches up with which #
14:13.57tzanger:-)
14:14.06*** join/#asterisk PBXtech (i=nik@229.sub-70-218-102.myvzw.com)
14:14.11tzangerwow rita's C4 now
14:14.36spackletzanger: headed for tejas - maybe
14:14.47tzangeryup
14:15.04znoGAriel_: ok, the dialplan for the sipura allows you to just change the dial tone when a certain number(s) is pressed, i was wondering whether the PAP2-NA can do this. Would be nicer I think than doing it on the Asterisk side, mainly cause I want a different dialtone if a "0" digit is pressed, i'd have to have an entry for "0" in Asterisk to do that r2 trick
14:15.19PBXtechone way audio is normally a nat issue right?
14:15.48Stephnie<PROTECTED>
14:16.06Ariel_znoG, the pap2-na has almost all the same setup as the sipura 2000
14:16.13tzangerok what is a PSLE system in terms of 911?
14:17.31*** join/#asterisk Dutts (n=dutts@81.168.70.41)
14:17.35Egonisast_merge_contexts_and_delete: Requested contexts didn't get merged
14:17.39EgonisWhy do I get this message?
14:18.13*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
14:18.58Duttshey everyone, I'm trying to compile * 1.0 from cvs, but get Makefile:100: *** missing separator. Stop. when I try to compile asterisk, zaptel and libpri works fine
14:19.00*** join/#asterisk zedkatuf (n=audela@62.231.155.4)
14:20.02tzangerPSLE = public safety and law enforcement apparently
14:20.14*** join/#asterisk NoNeo (n=Artur@193.24.24.10)
14:22.08*** join/#asterisk sphing (n=sphing@n128-227-48-31.xlate.ufl.edu)
14:22.52spackletzanger: Do you think it is pronounced as "puzzle"?
14:23.18*** join/#asterisk langals (n=icechat5@196.7.14.183)
14:25.11Duttsis voip-info down?
14:25.43littleballit seems down. i cnanot access also
14:26.06*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-194.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
14:26.44Duttsmy line 100 in my Makefile for * is ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk why would this cause a missing sepataror error with make?
14:26.44spackleDutts: no it's just a little sad
14:26.51Duttsspackle: =)
14:28.07azzie_Dutts: you're on linux?
14:28.15Duttsyup RH
14:29.05tzangerok listen up
14:29.14spackleyessir
14:29.21Dutts=)
14:29.21tzangereveryone in north america who wants to do VOIP 911 wants to get PSALI service
14:29.36*** join/#asterisk oej (n=Olle@64.251.114.2)
14:29.37Duttsazzie_: why you ask?
14:29.47azzie_Dutts: to make sure you're using GNU make ;)
14:30.07spackletzanger: and what is that => PSALI service?
14:30.12azzie_tzanger, what's PSALI ?
14:30.22tzangerPublic Service Automatic Location Identification
14:30.32tzangerCAD$2000 setup fee (Bell Canada)
14:30.36Duttsah.... so why iseas why this is happening? THis box already has an old CVS-HEAD installe don it and I wanted to use 1.0 instead so checked it out of cvs, everything buklts except asterisk witht his error..... don't get it
14:30.40tzanger$250 for 500 records
14:35.27*** join/#asterisk cianhughes (n=cian@cian.ws)
14:35.37Hmmhesaystheres a new wind blowing like i've never known, i'm breathing deeper than i've done
14:36.08spackleHmmhesays: what did you do with the money?
14:36.59Hmmhesaysyeah I wanna feel the sunshine, shining down on me and you, when you put your arms around me you let me know theres nothing in this world that I can't doooo
14:37.22*** join/#asterisk MikeJ[Laptop] (n=ircatjer@209-6-24-194.c3-0.sbo-ubr2.sbo.ma.cable.rcn.com)
14:37.35Hmmhesaysthat goes out to you niki, i hope you're happy in your solitude
14:37.38tzangerwow
14:37.45tzangerone wrong number yields ALL the 911 info I'll ever need
14:37.49tzangerfor beinga VOIP provider in Canada
14:38.31hypa7ianice
14:38.44tzangerbasically you want to subscribe to the PSALI service which lets you send ALI records (NANPA number + civic address) to the ILEC
14:38.59tzanger$2000 setup + $250 for 500 updates, min 2yr contract
14:39.06hypa7iathat's not too bad
14:39.14*** join/#asterisk scfrec (i=scfrec@scfrec.compic.ee)
14:39.21tzangerI also found out the correct way to call 911 for testing where you can't get the PSAP non-emerg # (or it doesn't exist)
14:39.28scfrechello, need help with h323 dial out
14:39.35tzangerwow
14:39.35langalsHI there...I am trying to install asterisk with speex - does anyone know how I can do this?
14:40.02tzangerBell Canada does not store addresses for DIDs, they do for BTN and LDNs but if you change a DID to an LDN you get a separate bill for it
14:40.09tzangerwhich is the whole point of this PSALI service
14:40.30mutilatordamned telco world and it's acronyms for everything
14:40.33tzangeryup
14:40.39scfrech323 provaider give me prefix in 000#0000 format and  format: Prefix+CC+Number
14:40.47scfrechow i can configure it in asterisk? h323 support added, g729 codec too
14:40.48spacklelangals: compile and install the speex library, then recompile asterisk.
14:40.49tzangerbut basically I can't send a call ot 911 and set CID+ANI to the DID because there's no address associated with it
14:40.55hypa7iayeah, that just made my brain explode
14:41.05Hmmhesaysthe guy from collective soul has such a great voice
14:41.14mutilatoruh
14:41.25mutilatorperty lips too eh Hmmhesays
14:41.41Hmmhesaysmutilator, i dunno are they?
14:41.53*** join/#asterisk NoNeo (n=Artur@193.24.24.10)
14:42.15scfrecanyone have time and can help ?
14:42.37Hmmhesaysi can probably help, but i'm hungover and kind of cranky
14:43.23Hmmhesaysmake that a lot cranky
14:44.07scfrech323 provaider give me prefix in 000#0000 format and  format: Prefix+CC+Number, how configure it?
14:44.24Hmmhesaysconfigure it where?
14:44.34Hmmhesaysdo you have an h?323 channel working right now
14:45.54scfrecHmmhesays: i call to asterisk with sip, but outgoing direction is h323
14:45.54olivier_dial(OH323/000#number@ipprovider)
14:45.58scfrechow to check it ?
14:46.13olivier_if you have oh323 of course
14:46.23Hmmhesaysso have you installed an h323 channel?
14:46.24scfreconly h323
14:46.46Hmmhesaysi cut my hand on a corona the other night
14:46.54tzangerHmmhesays: it's not twist-off, eh?
14:47.08Hmmhesaystzanger, i found that out the bloody way
14:47.13spackleHmmhesays: what were you doing so close to the sun?
14:47.27Hmmhesaysspackle, tanning those hard to get spots
14:47.31Hmmhesaysyou file
14:47.41Hmmhesaysi was an hour late for work this morning
14:47.46Hmmhesaysbut I really don't care
14:47.52Hmmhesayscause I hate this place
14:47.59filehaha
14:48.02spackleHmmhesays: hungover != slow
14:48.09olivier_are u sure you boss is not here ? ;-)
14:48.16Hmmhesaysnaw, i work pretty well hung over
14:48.18scfrecby moment call to 1000 and 500 is working. only need to do dialout to h323. h323.conf aviable, oh323.conf - not
14:48.38Hmmhesaysscfrec, so you have an h323 channel, installed?
14:48.54Hmmhesaysand I sing more
14:49.02spackleHmmhesays: what did you do with the money?
14:49.14Hmmhesaysspackle, wtf you talking about?
14:49.20scfrecHmmhesays: think yes
14:49.32spackleHmmhesays: the money your mom gave you for singing lessons.
14:49.46spackle;-)
14:50.03scfrecHmmhesays: how to check? from CLI for sample?
14:50.26Hmmhesaysscfrec, exten => _X.,1,DIAL(H323/000#0000${EXTEN}@<host>)
14:51.04Hmmhesaysspackle my parents were not supportive of anything musical
14:51.12Hmmhesaysthey were young and stupid
14:53.07*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
14:54.23FABRIZIOxxxhello all .. how can id do var = var + 1 in the dialplan?? I'm trying with exten => s,1,Setvar(count=0) exten => s,3,Setvar(count=$[${count}+1]) then exten => s,4,GotoIf(${count}=2?5,2) but its not working .. can anyone help please?
14:55.15scfrecNo channel type registered for 'H323'
14:55.21*** join/#asterisk [ViRii] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com)
14:58.14spacklescfrec, which h323 add-on did you use, where did it come from?
14:58.44Hmmhesaysscfrec install an h323 channel
14:59.09scfrecHmmhesays: pwlib + openh323
14:59.41spacklescfrec, you still need to install a channel to make use of those libraries
14:59.52Hmmhesaysscfrec you still need to install an h323 channel for asterisk
15:00.03maruzusing mnager i cannot put a call on hold, correct?
15:00.56mutilatorCS: 1.6 running on cs1.stonedinvaders.com and CS: Source running on cs2.stonedinvaders.com can anyone test either out for me?
15:01.34scfrecHmmhesays: trying...
15:01.46Hmmhesaysdon't try grasshopper
15:01.50Hmmhesaysjust do
15:01.59[ViRii]asterisk picks up on 2 rings? how would i go about changing that?
15:02.26shmaltzif I don't want to use digium hardware for PRI, what are my other options?
15:02.58Hmmhesaysi got a little change in my pocket going jing a ling a ling, won't you call me on the telephone baby give me a ring
15:04.39olivier_<shmaltz>sangoma
15:04.56olivier_but digium hardware is great
15:05.26shmaltzolivier_, yeah until you don't run into problems, like massive echo
15:08.13Hmmhesaysbut each time we talk I get the same ol' thing
15:08.29langalsspackle - the problem is that I am using ubuntu linux and installing from the repositories - I am trying to avoid installing asterisk from source
15:08.58*** join/#asterisk phpboy (i=shane@c1-79-1.tbnb.isadsl.co.za)
15:09.45phpboyWhat's an MSN number?
15:09.51phpboyand what does it stand for?
15:10.45langalsspackle - is it right that when asterisk is properly configured with speex, there should be a file called codec_speex.so in the asterisk modules directory?
15:10.48phpboy:/
15:11.30Hmmhesayslathos42, you know you like it
15:13.18wunderkinphpboy: http://www.eicon.com/support/helpweb/DIVA/MSN.HTM
15:13.27phpboyta
15:13.57*** join/#asterisk Nix (n=Nix@81.214.255.57)
15:14.40wunderkinthat backgrounddetect problem is driving me nuts ;/
15:17.44*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
15:19.03[ViRii]hey defaultly, asterisk picks up on 2 rings, with a digium card, how would i change that it picks up on the first ring?
15:20.23wunderkin[ViRii]: its waiting for callerid, you would have to turn that off
15:20.51[ViRii]thank you
15:21.53*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
15:22.15DrukenWrkcould be just me, but that second ring caller id thing is a pain in the ass, CID should be sent first imoho
15:23.11LoRezDrukenWrk: with some systems it is, but after a polarity reversal
15:23.34*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
15:24.14LoRezthe instruments "need" a way to know when to wake up and listen for the CID so they don't burn battery power if not needed.
15:24.15BhaalHey guys, got a small problem..  Ive just setup a small IVR .. but when I call in via a cell fone no tones are sent through..  Is this normal?
15:24.39LoRezno tones are sent or no tones are decoded?
15:25.02BhaalHey lorez
15:25.14LoRezdue to the compression of your cell circuit, it may be more difficult to recognize
15:25.18BhaalWell I cant hear any tones when I hit a digit and asterisk isnt registering anything...
15:25.54BhaalI cant hear them in the cell ear piece I mean
15:27.23BhaalJust out of question does asterisk have any voice recognition software?
15:27.38LoRezI've had my nokia set to not make noises locally, but that didn't affect the remote operation
15:27.49LoReznot afaik.  certainly not out of the tgz
15:28.00BhaalHrm... on both counts...
15:28.12wunderkinBhaal: no
15:28.31QwellFYI, newegg is the shit.
15:28.35LoRezBhaal: patches welcome?
15:28.36DrukenWrkBhaal: it's possible... but a royal pain in the ass...
15:28.36BhaalOkey, can asterisk be told to echo back the tones when a digit is pressed?
15:28.46[ViRii]how do i use the intercom feature with polycom 600 phones and asterisk?
15:28.59QwellI buy something yesterday morning, I pay for regular fedex saver shipping, and I'm getting my stuff today at around noon
15:29.17filedon't get me started on shipping
15:29.31DrukenWrkQwell: perhaps they shipped it from your city?
15:29.38fileDHL = evil
15:30.01QwellDrukenWrk: about 50 miles away
15:30.17Qwellbut hey, $7 for "overnight" shipping...thats not bad
15:30.19DrukenWrkfile: i can't agree with you more... DHL or better yet, LOOMIS, sucks ass
15:30.54DrukenWrkQwell: yeah not bad... i don't do overnight, so i dunno... but i do sameday for 15...
15:31.56Qwellits not real overnight.  overnight was like $30
15:32.24DrukenWrkouch
15:32.32DrukenWrkfor what size/weight?
15:32.47Qwellall I have to do is pay the $3.99 rush processing fee, and I can almost guarantee it to be here by noon the next day
15:32.57Qwelldunno, a 4 pound 8 port switch
15:33.12brettnemer nose
15:33.15Qwellbrettnem: still a cat 3?
15:33.30brettnemit's a 4 now..
15:33.37Qwelluh oh
15:33.42Qwellstill headed for TX?
15:33.45brettnemI think they are expecting it to shift from a 4 to a 5 and back for a while
15:33.50brettnemoh yeah, no way out at this point
15:33.56spackleBrettnem, are you in Tejas?
15:33.59brettnemyep
15:34.07brettnemright smack dab in the middle
15:34.08spacklewhat city?
15:34.12brettnemAustin
15:34.13WonkaDHL is Deutsche Post, of course they suck
15:34.23*** join/#asterisk kpettit (n=keith@69.15.174.114)
15:34.28Qwellbrettnem: a bit far from the coast though, right?
15:34.28Wonka<- knows. lives in .de
15:34.42brettnemQwell: yeah, but I have quite a bit of family in Houston
15:34.49brettnemwhich is only 60 miles from the coast
15:34.56Qwellahh...
15:35.05spackleand actually has a sea port
15:35.12*** join/#asterisk oej (n=Olle@64.251.114.2)
15:35.14brettnemwell.. galveston.. yes
15:35.36spackleoh, I think you just jinxed them
15:35.57brettnemI grew up in Houston.. in 83 we had a cat 3 hurricane that knocked over a 4 ft wide willow tree only 6 feet from my back door
15:36.09brettnemheh galveston? nah, they've always been jinxed
15:36.15*** join/#asterisk jief- (n=jief@modemcable163.182-80-70.mc.videotron.ca)
15:36.29Qwellbrettnem: have they started a warning yet or anything?
15:36.41brettnemheh nope.. they don't do that until it's too late..
15:36.56Qwellahh, heh
15:37.12brettnemthey say it may hit anywhere from Louisiana to Northern Mexico.. that's a really really big span.. could be like 800 miles
15:37.19brettnemnah maybe 600
15:37.30*** join/#asterisk [iL]TraXo (n=TraXo@vs250211.vserver.de)
15:37.41Duttsdoes anyone here have snedmail installed ond configured on their * boxes? or would recommend another mail system? All I basically wantr to do is allow Asterisk's voicemail to email me, don't need the box to do anythign else
15:37.42jief-ok, we have a working * pbx at the office. but i'd like to setup direct inward calling, so if people call 555-1234, it goes directly to an extension. but the problem is, all my calls go to [incoming], and there's an 's' extension there, so i can'
15:37.50jief-t match using 5551234
15:37.56brettnemDutts: just use sendmail for that..
15:38.11brettnemjief-: where is the call coming from? Technology?
15:38.20spackleDutts, depending on your distro there are only one or two params to change.
15:38.29Duttsbrettnem: never managed to get the hang of configuring sendmail, I'm using RH8
15:38.44jief-brettnem: my telco, PSTN line
15:38.47brettnemDutts: depending on your provider, you might not need to do anything to make sendmail work.
15:38.51brettnemjief-: POTS line?
15:38.54jief-brettnem: yes
15:39.00Duttsbrettnem: how can I test it?
15:39.06brettnemjief-: How would the computer know what phone number was dialed?
15:39.31Qwellunless its a DID line, or whatever
15:39.35brettnemsendmail -s "some crazy subject" youremail@goeshere.com
15:39.45brettnemthen type some text.. <enter> type ONE dot <enter>
15:39.55jief-brettnem: well, i know its possible with T1s because of DIDs, not sure if that works with POTS
15:39.58fileACHOO!
15:40.04*** join/#asterisk ezio (n=ezio@host19-96.pool8289.interbusiness.it)
15:40.07brettnemjief-: are these DID trunks?
15:40.16BhaalWhats the easiest way to dial a whole bunch of extensions at the same time and give the incoming call to whoever picks up?
15:40.33QwellBhaal: SIP/bob&SIP/susie&SIP/jeff
15:40.41spackleBhaal, always the same extensions?
15:40.46Bhaalspackle: yeah
15:40.49[ViRii]hey how do i use the intercom feature?
15:40.54Qwellif they're always the same, mine will work.  if not, queues maybe?
15:40.55spacklelike Qwell says then.
15:40.59BhaalQwell: ahh right...
15:41.01jief-brettnem: i would have to check with my provider i guess
15:41.02brettnemintercom feature?
15:41.07jief-brettnem: im very new to telephony
15:41.11brettnemjief-: is it a regular phone line?
15:41.11*** part/#asterisk ezio (n=ezio@host19-96.pool8289.interbusiness.it)
15:41.21jief-brettnem: its a regular business line
15:41.22Qwell[ViRii]: usually you setup a line that will automatically answer, and you register that line, and the normal line
15:41.26szerbye
15:41.29Duttsbrettnem: ok tried that but nothign in my inbox, are there any lgos I can look at to see what went wrong?
15:41.47brettnemjief-: then it probably isn't a DID. which is why it goes to an 's' extension. no way to tell what number was dialed
15:42.21brettnemDutts: depending on your distro maybe something like /var/log/maillog
15:42.26wunderkinjief-: i think to do that you would need to put each port into its own context
15:42.37brettnemwunderkin: port?
15:42.45[ViRii]thank you
15:42.53filebrettnem: you're being too helpful
15:42.55brettnemoh FXO port.. right
15:43.00brettnemfile: I'm sending them bills
15:43.01jief-wunderkin: i could do that, but i cant, because all lines are part of a pilot.
15:43.03brettnem:)
15:43.17brettnempilot?
15:43.27jief-brettnem: cascade?
15:43.37jief-several lines for 1 number
15:43.39brettnemjief-: hunt group?
15:43.40brettnemah
15:43.55jief-brettnem: as far as ive seen, different telcos call this in various ways
15:44.00brettnemhmm well maybe you have a DID trunk.. I haven't used those in years..
15:44.15brettnemjief-: especially depending on what country you are in
15:44.31*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
15:44.41brettnemok, everyone give me your address so I can send out invoices ;)
15:44.42jief-brettnem: looks like ill have to check with my provider
15:44.52spackle~seen lathos42
15:44.56jbotlathos42 is currently on #asterisk (3h 29m 41s).  Has said a total of 1 messages.  Is idling for 3h 29m 27s
15:45.07brettnem~seen jobot
15:45.10jbotjobot <~chatzilla@dpc6935220107.direcpc.com> was last seen on IRC in channel #familiar, 384d 11h 48m 32s ago, saying: 'got any bootblaster gurus on tonight?'.
15:45.10brettnem~seen jbot
15:45.12jbotjbot is currently on #asterisk-doc (13d 1h 21m 14s) #storm (13d 1h 21m 14s) #uphpu (13d 1h 21m 14s) #how (13d 1h 21m 14s) #ol (13d 1h 21m 14s) #flyspray (13d 1h 21m 14s) #phlyte (13d 1h 21m 14s) #asterisk (13d 1h 21m 14s) #byumug (13d 1h 21m 14s) #tacobeam (13d 1h 21m 14s) #aegis ...
15:45.22brettnemhmm busy bot
15:46.04brettnemtacobeam?
15:46.13jief-brettnem: my question is, when i get a call on my digium card, it sends them to [incoming]. in that context, i have no choice but to have s,1,Answer. I just dunno, how in [incoming] i could match, 551234, to dial SIP/100. because it will always go to 's'
15:46.27tzangerjief-: which card?
15:46.41Duttsbrettnem: nah it;'s not configured, I get User unknown from localhost.localdomain, where do I configure it to set my ISP's SMTP server?
15:46.56*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
15:47.22brettnemjief-: what happens if you do exten => 551234,1,Dial(SIP/100) or exten => s,1,Dial(SIP/100)
15:47.24spackleDutts, do you want to send it to ISP server or internal server?
15:47.38jief-ok, i guess ill have to post my dialplan somewhere
15:47.40Duttsspackle: ISP's server ideally, as I don't have an internal server
15:47.44tzangerjief-: what card?
15:47.48brettnemhmm.. Dutts google "smart host sendmail" without the quotes
15:47.52jief-tzanger: we have a digium with 4 fxo ports
15:47.57tzangeryou won't get it with that
15:47.58brettnem~pastebin
15:47.59jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
15:48.14tzangera normal POTS line does NOT tell you what # was called
15:48.21tzangeryou can get caller ID but that's not what you're looking for
15:48.46brettnemtzanger: I don't know if he knows if he has a normal line
15:48.48tzangerFXO and FXS ports can only get to the 's' context -- there is no DID capability there
15:48.56brettnemoh,, well there ya go
15:48.57tzangerbrettnem: the FXO ports don't support analog DID IIRC
15:49.05jief-tzanger: ok, that's what i thought from the beginning, and my boss argued about it. now, is it possible to have my provider enable that on POTS lines? or is it only available with PRIs?
15:49.09spackleDutts: you configured sendmail before?
15:49.32filewhatever happened to our love? I wish I understood
15:49.37tzangerjief-: only available on PRI as far as I'm aware.  htere might be some goofy hack-up available but the chance that Digium's wctdm driver supports it are somewhere between slim and none
15:49.47brettnemjief-: DID lines are a completely different beast..and I'm not sure of their availabilty over POTS
15:49.49*** join/#asterisk santiago (n=santiago@63.245.86.145)
15:49.51Darwin35on some home likes you can dial *5 and get the last nmbr dialed from your phone
15:50.02spacklefile, are you listening to ABBA 24/7 or what?
15:50.03jief-tzanger: thank you. my headache is gone now ;)
15:50.03brettnemtzafrir: I'm pretty sure you can do it over T1 DID.. like FGD or something weird like that.
15:50.08tzangerDarwin35: that's redial.  again, that's not DID
15:50.10Darwin35I have it setup to record the last nmbr dialed
15:50.10filespackle: A-Teens actually
15:50.21Ahrimanesfile: omg
15:50.29Darwin35no it just says the nmbr not redials
15:50.47BhaalOkey, how do I let call ring for like 2 seconds before answering?
15:50.48tzangerDarwin35: again, that's the last # you called out, not hte number that was called coming in
15:50.54Bhaaljust Wait(2) ?
15:51.00tzangerjief-: you *can* have both ports go to different contexts
15:51.11tzangeri.e. port 1 you KNOW they dialed 5551234 and port 2 you KNOW they dialed 5551235
15:51.21tzanger(assuming no dring or call forwarding)
15:51.32tzangerand you might be able to get dring to work if you subscribe to it
15:52.08brettnemwhy is there an entry for "ss7" in my zapata.conf.sample file?!
15:52.12brettnemwtf
15:52.23filethat's super sekret
15:52.24tzangerbrettnem: to give ya hope :-)
15:52.38brettnemyeah WHATEVER
15:52.51brettnemprobably F-Link baloney. ;)
15:52.58Darwin35e911 is a pipe dream full of hash
15:53.04brettnemit's PRI in sheep's clothing
15:53.05Darwin35light and inhale
15:53.13brettnemDarwin35: e911 isn't a pipe dream?!
15:53.17tzangerDarwin35: yeah but I now know how to get my data into the ALI database
15:53.34brettnemeasy enough
15:53.41tzangerso not quite e911 but I can set my outgoing CID+ANI to the DID and get the right address to come up with 911's dialed
15:53.57Darwin35hmm
15:54.06brettnemI've done quite a bit of 911 interop with asterisk
15:54.10Darwin35do you have a page on how to do this . it would helkp many out
15:54.30brettnemthere isn't any particular magic to it.. what part are you having trouble with?
15:54.34Darwin35I know that we have ti implamentate it soon
15:54.48Darwin35we have not even starte yet
15:55.03Darwin35I know its going to be one of the things on the new job I have to do
15:55.10brettnemyou haven't starte ti implamentate?
15:55.25brettnemDarwin35: are you with a CLEC?
15:55.44brettnemDarwin35: are you italian? I have to ask.. ;)
15:55.47Darwin35I just started and no we have a statement we dont offer e911 right now but we will before the end of the year
15:55.54Darwin35no
15:56.04brettnemSo what's your plan? :)
15:56.07Darwin35Canadian living and working in the US
15:56.11brettnemah
15:56.15brettnemabove the law
15:56.20brettnemor at least.. the US
15:56.50brettneme911 isn't magic.. what part are you confused with?
15:57.08Darwin35I have yet to see a basic setup out there to start with
15:57.10brettnemi2, the new FCC recommended 911 method for VoIP will be magic..
15:57.22Darwin35not even a paper with a baisc outline for asterisk
15:57.30brettnemok, if you can't describe the problem, I can't offer a suggestion.
15:57.47tzangerbrettnem: nah Bell Canada has a PSALI service you use
15:57.59tzanger$2000 setup, $250/mo for 500 changes, min 2yr contract
15:58.00brettnemyeah, so? PSALI works in similar ways
15:58.03Darwin35I need to see a basic setup.
15:58.17brettnemjust use dash911 for those rates.
15:58.32Darwin35to grep and understand the database calls and passing the info to the 911 service correctly
15:58.34brettnemHmm.. I don;'t know if I've ever recieved a bill from Intrado..
15:58.49brettnemthere arn't databae calls to pass info to the 911 service
15:58.52tzangeryeah -- you can do that, or you can redirect 911 to your own primary PSAP centre and have them look up your private DB and route to the correct secondary PSAP
15:59.04brettnemyuck.. don't do that
15:59.12tzangerbrettnem: why not?
15:59.20spackleits a PSAP smear
15:59.21brettnemthen you'll have to manage MSAG and ESN/PSAP routing..
15:59.50tzangerbrettnem: true but if you're a small provider with a relatively local customer area it's a lot cheaper and works fine
15:59.59brettnemget a connection to a CLEC and get PS/ALI or get your own 911 trunks to the incumbant route selector.. unless your service area is the size of a pinhead
16:00.13Darwin35we are a national provider all of the US
16:00.15Darwin35brb fone
16:00.36tzangeryes typically you do PSALI to your LEC ... if you're big enough to hook on to the PSAP tandems you typically don't need to worry about any of this anyway :-)
16:00.39brettnemtzanger: well, you better be sure all of your customers actually need all of the EMS, Fire, AND Police to that particular PSAP
16:00.53tzangereh?
16:01.21brettnemsure you do.. I'm hook up to the LEC PSAP tandem (they call this device a 'route selector') and I still hve to send NENA formatted records to the ALI DB.
16:01.37tzangerhmm
16:01.45tzangerI'm starting to drown in the alphabet soup
16:01.55*** join/#asterisk klictel (n=klictel@207.107.208.140)
16:02.18brettnemALI records get validated with MSAG which returns a ESN. An ESN is a particular grouping of Police, Fire, and EMS. A PSAP is responsible for a list of ESNs.
16:02.37brettnemThis is because sometimes you cross the street and it's now under the jurisdiction of a different police department, but it's the same fire and EMS
16:03.22brettnemif you do PS/ALI with a CLEC, the LEC does the MSAG lookup and the ESN routing to the approprite PSAP.. if you don't do that. .and get CAMA trunks for example to the PSAP, you have to be sure you are sending the call to the appropriate entity..
16:03.29jief-tza/quit
16:03.32sivanaIt's so hard to soar with the eagles; when you're surrounded by turkeys
16:03.34brettnemif your customer serving area is tiny, this might not matter.
16:03.38BhaalWhy is voip-info so slow sometimes?  Lack of bandwidth or CPU?
16:03.51brettnem330 turkeys all hitting it at the same time
16:04.03sivanahehe
16:04.43Duttsdoes anyone have a sendmail.mc for RH8 that uses a smarthost to send all mail via their ISPs SMTP server? I am having trouble finding an example on the net that works
16:04.50brettnem911 isn't magic.. but because of all of the seperate jurisdictions involved  and city governments, it can be confusing.. technically it's quite simple.
16:04.51spackleAnybody do much videoconferencing (or support of) is h.323 the golden standard?
16:05.05spackleDutts: I was about ready to help you when you disapeared
16:05.07azzie_Dutts: just fix it in sendmail.cf
16:05.08brettnemDutts, smarthost is pretty standard..
16:05.18sivanaA Cama is a hybrid between a camel and a llama
16:05.21brettnemyeah, I wouldn't worry about the whole .mc baloney
16:05.23*** join/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-206.rockynet.com)
16:05.38azzie_Dutts: DS entry
16:05.40Duttsspackle: sorry I googled for smarthost and got tons of examples, trie one here http://www.elandsys.com/resources/sendmail/smarthost.html but it doesn't work
16:05.41brettnemisn't a camel just a lama with  hump?
16:05.50sivanagood question
16:05.57brettnemdid you restart sendmail?
16:06.31Duttsno I get errors using m4 with that mc file
16:06.32azzie_Dutts: put DSmail.provider.com in your sendmail.cf and restart sendmail
16:06.36sivanabrettnem: our ILEC wants $2400/mo for a 911 route the PSAPs
16:06.51brettnemif you changed it in .mc I think you have to do some sort of m4 recomplie
16:07.01brettnemsivana: WHAT?!
16:07.08sivanayup
16:07.18brettnemsivana: how are you connecting to them? what is your relationship? carrier? user? what?
16:07.25sivanaI'm not
16:07.42brettnemyour not.. what?? telling??
16:07.45sivanaI'm not using the ILEC at this point.. we have a CLEC here that are providing non-911able PRIs
16:07.53Duttswhere abouts in sendmail.cf do I nbeed to put that?
16:07.58brettnemsivana: where is here?
16:08.03sivanaNorth Bay
16:08.06sivanaOntario, Canada
16:08.09azzie_Dutts: LOOK FOR "DS" LINE IN YOUR SENDMAIL.CF
16:08.32Duttsazzie_: as yes found it, so I just add my ISP's smtp server in there?
16:08.41azzie_Dutts: YES!!!
16:08.58spackleand restart sendmail
16:09.13brettnemazzie_: look for your "Caps Lock" key and please depress it
16:09.23Duttsazzie_: sorry dude! =) hehehehehehe any particular syntax? or is 'DS smtp.eclipse.co.uk' (without the quotes) ok?
16:09.40sivanatzanger: ping
16:09.50BhaalQwell: Okey, I just tried it dialing multiple extensions on an incoming call...  Seems to be ringing them individually and waiting for them ring out before continuing to the next extension...  Is it possible to make them all ring at the same time?
16:09.53spacklemake sure the quotes are exact ` and ' just like ofther entries
16:10.00azzie_brettnem: this guy seems to understand better with caps, sorry anyway :)
16:10.17brettnemsivana: as a carrier, I think I pay like $35 per 911 trunk which is actually refunded to me by the 911 agency.
16:10.21azzie_Dutts: yes, and no spaces either
16:10.38brettnemDutts: read: no spaces AT ALL <-- notice cap usage
16:10.39sivanabrettnem: where are you located?
16:10.41azzie_Dutts: what's this to do with asterisk anyway?
16:10.44brettnemsivana: let me check
16:11.01brettnemI'm in Austin,Tx
16:11.07brettnemjust waiting for the storm
16:11.10sivanaya, I think the US system is different
16:11.12spacklebrettnem, you can't really see it yet can you?
16:11.15Dutts=) sorry guys, all I want is to allow my voicemail to be emailed to me..... got the voicemail all set up, but sendmail don't work, google searching finds about a billion different guides on sendmail, none of them do what I want to do
16:11.21sivanaBell Canada owns the network here and it's "private"
16:11.28DuttsI can't believe if it's just a one liner that it isn't somewhere mroe obvious
16:11.39brettnemspackle: nah, it might not even get here..
16:11.48brettnemDutts: sendmail config is very obtuse
16:11.56brettnemsivana: private networks suck
16:11.58spackleaustin=bats or san antonio=bats?
16:12.05sivanayup.. it's bs
16:12.06brettnemspackle: austin=bats
16:12.22tzangerBhaal: easily:  Dial(SIP/100&SIP/200&SIP/300)
16:12.25tzangerworks perfectly fine
16:12.45Duttsbrettnem: yeah I can tell from all the possibilities in there!
16:13.01sigtermseems to be a common thing latey, waiting for storms
16:13.01tzangerBhaal: I strongly suggest reading the asterisk handbook draft
16:13.04tzanger~handbook
16:13.05jbothandbook is, like, http://www.digium.com/handbook-draft.pdf
16:13.05brettnemspackle: I was driving down the freeway the other day and I saw this weird black cloud spiraling out of the underside of the freeway in this huge spiral into the sky.. as I got close I notices it was a huge trail of bats.. pretty cool!
16:13.35*** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk)
16:13.41*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0ih.dialup.mindspring.com)
16:13.53Bhaaltzanger: I just tried that exact example...  Ahhh wait, I had a time in there... doh
16:14.38lenne_dkGoodevening channel. Is there somewhere to submit patches and bugreports?
16:14.48Bhaaltzanger: Can I put Dial(SIP/100&SIP/200,60)  And it will wait for 60secs on both?
16:14.49tzangeryou can still do it with timeouts
16:14.52tzangerit will ring them all for 60s
16:14.54spacklebugs.digium.com
16:14.56Bhaalyep
16:14.58Bhaalthanks
16:15.00tzangerwhoever picks up first stops the ringing on the rst
16:15.39Hmmhesayscause we were barely 17 and we were barely dressed
16:15.42Duttsok guys so I've restarted sendmail after adding that line, if I do 'sendmail -s myemailaddress', type some text, then hit . on a new line, it ends up appearing in my /var/log/mail/root?
16:17.51*** join/#asterisk ManxPower (n=eric@ppp-70-246-171-89.dsl.lgvwtx.swbell.net)
16:18.16Duttsah sorry syntax incorrect..... that defaults to root.... got a new problem now, they ar egetting bounced back from my isps smtp server with Sender address rejected: Domain not found, at the moment it's still coming from root@localhost.localdomain,
16:18.55spackleDutts, redhat has another line you have to adjust to send mail off the machine.
16:19.35Duttsspackle: ah right, sorry, me rushing ahead... do you know what that is?
16:19.56azzie_Dutts: go back to your sendmail.cf to the line Dj and make it DjMyHost.MyDomain.Com <- replace with REAL hostname
16:21.41brettnemalso, sendmail syntax is sendmail -s "subject line" emailaddress@example.com
16:22.06brettnemwow.. so my whole family from houston is coming up with.. (counting) 9.. WTF. NINE dogs
16:22.10spackleI know, the thought of all those glorious voicemails coming through my email gets me all twitterpated too.
16:22.21Beirdo9 dogs?!
16:22.23BeirdoWTF?
16:22.26spacklelucky day
16:22.39brettnemyeah
16:22.51brettnemand I've got 2.. great.. it's like noah's freakin arc
16:23.08Beirdoa lot of buttsniffing goin on
16:23.27brettnemyou mean, the dogs?
16:23.32Beirdohopefully
16:23.33Beirdoheh
16:23.45Beirdoyou said Houston, not Little Rock
16:24.27brettnemhaha
16:24.48Hmmhesaysfraggle rock?
16:25.42Wonkabrettnem: one STANAG mag later, there are several dogs less...
16:26.03brettnemSTANAG?
16:26.24kpettitanybody know how to get ntpd to work with polycom phones?
16:26.45brettnemyeah, log in, put in a valid sntp address that can resolve with DNS setup in the phone
16:26.48fiber0ptiDoes anyone have any experience with Time Warner Telecom PRI's and asterisk?
16:27.01kpettitso going by IP won't work?
16:27.11BeirdoI hate computers
16:27.14brettnemfiber0pti: I do
16:27.26azzie_fiber0pti, pretty standard PRI i'd say
16:27.32brettnemkpettit: sure it should.. but the phone needs internet access if it's hitting one across the internet
16:27.33kpettitbrettnem, I've been putting the asterix box as the ntp server, and the service is running.
16:28.04kpettitthat's what the issues is, it works across the internet but not to the local ntp server.  but I can do "ntpdate servername" from a regular laptop and it works just fine
16:28.08brettnemkpettit: that should work.. might be a permissions problem in the ntpd.conf file..
16:28.19fiber0ptibrettnem: I'm trying to figure out how to interface with a digium card with the equipment time warner provides. Specifically do I need to purchase any special module for their equipment?
16:28.29Daminkpettit: Or perhaps a firewall rule gone wild..
16:28.34brettnemfiber0pti: a pri is a pri is a pri is a pri..
16:28.37spacklebrettnem = johhny-on-the-spot today
16:28.37kpettitI think it's a issue with the poloycom phones from what I've been reading, just don't know if there's a way I can get around it
16:28.40sivanadoes anyone know if date/time is sent out the fsk?
16:28.49brettnemspackle: slow day. dont' tell
16:29.00sivanacustomers are saying the time is wrong on our system when calls come in
16:29.10brettnemsivana: I think so.. I've never set the time on a CID box.. <shrug>
16:29.16kpettitthe asterisk box acts as a firewall for the phones, and it can get ntp from outside, but not from the localntp server.  It's really weird
16:29.19brettnemsivana: check the timezone on the box and the actual time
16:29.20sivanabrettnem: it's the * box
16:29.24*** join/#asterisk bumblefsck (n=bumblefs@69-160-158-193.ontrca.adelphia.net)
16:29.46brettnemthen there is most likely someting wron with the local ntp server
16:30.03brettnemfiber0pti: really a time warner PRI is about as standard as you'll get.. just plug the sucker in
16:30.14sivanabrettnem: date on the * box is correct.. unless it's the linksys adapters changing on the customer end
16:30.21brettnemsivana: start with a netstat -nlp to see if it's listening for ntp requests
16:30.34brettnemsivana: check the timezone on the phone and on the computer
16:30.42brettnemwait
16:30.56brettnemis the problem that the time is wrong, or that it doesn't get time at all?
16:32.08sivanatime is wrong on the customers' phones
16:32.11*** join/#asterisk [Lamer] (i=Lamer@221.128.103.229)
16:32.21sivanawe deploy linksys RT31's to them
16:32.29brettnemthat shouldn't matter
16:32.31[Lamer]Hi, there can I hang up a zap channel from CLI?
16:32.53brettnemsivana: so it is getting the time from the ntp server and the timezone is just off? or does it totally fail to communicate with the ntp server
16:32.59sivanaI didn't think * was sending out time, but I've a couple of customers tell me the same thing
16:33.13brettnem[Lamer]: try soft hangup Zap/1-1 or whatever
16:33.14phpboyexten => s,1,Dial(SIP/10,20,tr)
16:33.21[Lamer]I often recieved unidentified rings to ZAP channels
16:33.28brettnemsivana: * does NOT send out time
16:33.33phpboydoes that look right for receiving a call from an ISDN card?
16:33.36sivanaI thin the timezone is off.. I believe the linksys ATAs are manipulating it
16:33.46[Lamer]brettnem: from CLI?
16:33.47brettnemsivana: well except in FKS CID
16:33.51brettnem[Lamer] yes
16:33.52sivanayes
16:33.56sivanathat's the part
16:33.59brettnemah
16:34.08brettnemso is the minute right whene they get it?
16:34.10fiber0ptibrettnem: via an RJ45 connection?
16:34.11sivanabut "date" on the * server is correct
16:34.15phpboyI would just like to start off by saying... I love you guys
16:34.17phpboy!!!
16:34.35sivanabrettnem: not sure.. I think the RT is changing the FSK
16:34.38brettnemfiber0pti: yep.. pretty normal stuff.. it's a T1 in PRI's clothing
16:34.43sivanaI will test it
16:34.54brettnemRT?
16:34.57[Lamer]brettnem: thanks, that really works
16:35.03sivanabrettnem: linksys
16:35.11brettnemgood.. btw, you shouldn't get phantom rings!
16:35.22brettnemsivana: you think the router is messing with your RTP?
16:35.45sivanasivana: the linksys RT31 have the phone ports built in
16:35.48brettnemsivana: I find that to e unlikely
16:35.52brettnemoh I see..
16:36.09sivanaone of them, either * or the RT31 is manipulating the FSK clock stamp
16:36.11brettnemI'm not totally sure if time is sent in the FKS signal.
16:36.20brettnemyou might need to setup ntp in the ATA
16:36.24sivanaya
16:36.24[Lamer]brettnem: yeah, I'm trying to find the cause of those <virtual> rings
16:36.35brettnem[Lamer]: what protocol are you using?
16:36.41sivanaI think the ATA does have time settings that up until now, I've ignored :)
16:36.46phpboydoes that look right?
16:36.48phpboyexten => s,1,Dial(SIP/10,20,tr)
16:36.48[Lamer]brettnem: SIP
16:36.50brettnemsivana: try it.. let us know
16:37.01brettnem[Lamer] Ithought you said your calls were ZAP?
16:37.02sivanafor sure
16:37.05spacklesipura ATAs do have a time setting.
16:37.10[Lamer]brettnem: oh, yeah
16:37.29brettnemphpboy: that line connects the channel to a SIP phone named "10" it has nothing to do with the incoming channel
16:37.33[Lamer]brettnem: ZAP on the X100P clone
16:37.44brettnem[Lamer]: ok, so it's not SIP
16:37.59brettnem[Lamer]: so phantom rings on a regular POTS line eh?
16:38.04BhaalCan asterisk play .wav files for an IVR instead of .gsm files?
16:38.10[Lamer]brettnem: yes, you're right
16:38.11brettnemcould be some kind of VM notification.. maybe..
16:38.18spacklebingo
16:38.23brettnemBhaal: yes, if they are formatted right.
16:38.28brettnembingo? you won?
16:38.40phpboybrettnem: incomming from pstn...
16:38.43spackleno you did jhonny-on-the-spot
16:39.00spacklethat's some serious nervous energy you have going today.
16:39.23*** join/#asterisk rd1966 (n=rd1966@82-70-97-100.dsl.in-addr.zen.co.uk)
16:39.24brettnemBhaal: playback will attempt to pick the best audio file to match the channel.. wav files have to be formatted in a valid format (ulaw, gsm, etc)
16:39.27brettnemme?? heh
16:40.12[Lamer]brettnem: any idea how I can avoid these rings?
16:40.14Bhaalbrettnem: Oh, how do I do that?
16:40.42brettnem[Lamer]: unplug your phone line.. ha.. sorry. no really.. what I'd do is plug a real phoen into it and see if it rings too.. if it does, call the phone company and ask them why it's doing it.
16:40.50NORANDOMSi sure love me some spackle
16:40.58brettnemBhaal: how do you do what?
16:41.29[Lamer]brettnem: I tried that it didn't ring
16:41.31brettnemBhaal: when you record the file, just record it in one of those formats.
16:41.37brettnem[Lamer]: is it predictable?
16:41.57brettnemshow application record
16:42.06[Lamer]brettnem: it happened after I made a call to the ZAP channel
16:42.17brettnemif you need to convert an existing file, use sox. please don't ask me how to use sox. go read about it.
16:42.29brettnemyou make a call, hangup and then you get a ringback?
16:42.46[Lamer]brettnem: yes
16:42.56brettnemwhat signalling are you using?
16:43.02brettnemfrom zapata.conf
16:43.48Bhaalbrettnem: That would make it a gsm/ulaw format..  Not wav :)  Tis cool.. I will work that out in the morning :)
16:43.53[Lamer]brettnem: fxs_ks on my two ZAP channels
16:44.28brettnemBhaal: wav is an extension, there are lots of ways to encode wavs
16:45.06BhaalHrm, true...  I assume .wav to be a Microsoft audio format, thats all..
16:45.08*** join/#asterisk n0rf- (n=n0rf@ip87.66.1311D-CUD12K-03.ish.de)
16:45.24Bhaalbrettnem: Rather then just raw audio
16:45.58*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)
16:46.57*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:47.05brettnemBhaal: running windows.. start->run->sndrec32<enter> File->save as (look down at the bottom of the dialog) hit the "Change" button and examine the formats
16:47.06BhaalAnyway, nearly 3am.. off to bed for me...
16:47.50Bhaalbrettnem: Im using linux...  and I record most audio with Audacity.. I can only save out as wav/ogg/mp3 :)  I will work out converting them with sox tomorrow...
16:48.07brettnemok, when you save to wav, you should get a format option
16:48.40Bhaalbrettnem: Not with Audacity
16:49.21Duttsguys, just wanted to say thanks for helping me out with my sendmail problem..... it;'s working now =) THANKS!!!
16:49.22brettnemin audacity, see bottom left
16:49.26brettnemyou'll see the rate
16:49.30Bhaalbrettnem: But there are options I see in the preferences... all good
16:49.53brettnemactually on the channel you'll see rate, and format
16:49.55Bhaalbrettnem: Thanks
16:50.24Bhaalyah...  Tis all good...
16:50.32brettnempreferences->file formats.. it's there..
16:50.36Bhaalyeah
16:50.51brettnemand PCM will work for ULAW
16:51.07brettnemok, I'm done for now.. back to work
16:51.14Bhaaland asterisk decides based on filename extension?
16:51.20brettnemand channel codec
16:51.24*** join/#asterisk HeadachesAbound (i=user76@wsip-68-99-73-32.tu.ok.cox.net)
16:51.29BhaalAhh so it reads the file header cool
16:51.38brettnemso if channel is running ulaw, it'll prefer a .ul file over a .gsm file
16:51.41HeadachesAboundMorning everyone.
16:51.42brettnemno nono
16:51.47brettnemit reads the extension
16:52.01Bhaalbrettnem: Well its all g729 for me...
16:52.15HeadachesAboundDo the polycoms support md5 auth for sip registrations?  if so, what is required to make it work?
16:52.21Bhaalusing ATA's and my SIP <-> pstn provider uses g729
16:52.25brettnemso make your recordings in g.729
16:52.30brettnemwhat ATA?
16:52.46BhaalA NetComm V100LS and a Netgear somethingorother
16:53.10BhaalGo, do your work...
16:53.14arguilebrettnem: The SPA200x models?
16:53.16BhaalIm sleeping
16:53.16Bhaal:)
16:53.19brettnemarguile: yes
16:53.24BhaalThanks for the help brettnem
16:53.25brettnemg'night
16:53.30brettnemno prob
16:53.44brettnemarguile: they are very nice.. I'd recommend them..
16:53.53brettnemI think the 2100 will be able to do 2x g.729
16:54.03brettnemI can only do one at a time.
16:54.30azzie_brettnem: didn't ata186  had a similar limitation?
16:54.39brettnemI'd recommend a sipura ata over a low end sip phone any day
16:54.42*** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com)
16:54.55brettnemazzie_: the ata18 had all sorts of problems..
16:54.59brettnem+6
16:55.12KattyHeadachesAbound: hi.
16:55.16azzie_brettnem: i remember it could not do 2 calls with some codec ...
16:55.41brettnemcouldn't handle reinvites either
16:56.09arguileThe problem with the 2100 is that you're paying for the router/QoS instead of just the two FSX ports
16:56.26Hmmhesayswiki wild wild
16:56.33HmmhesaysKatty: I sang karaoke last night
16:56.43Hmmhesaysat the house of rock
16:56.57Hmmhesayshttp://www.playmakersfargo.com
16:57.25Hmmhesayswho are you
16:57.25*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
16:57.37brettnemhaha
16:57.40Hmmhesaysand give me a reason not to challenge you to a duel right now
16:57.48*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
16:58.29HeadachesAboundBecause I can fubar your machine with a DOS attack and happily report you to the RIAA for letting me 'evaluate' your musical montage
16:58.39*** join/#asterisk litage (n=nick@203.201.96.182)
16:58.42Hmmhesaysstuff it darth
16:59.00HeadachesAboundcan't.  working.
16:59.05brettnemhah
16:59.08arguileThe SOHO hardware I'd really like to get is the linksys WRTP54G which has a sipura 2 port FXS build in
16:59.18Hmmhesaysthey let you IRC while driving the bus?
16:59.24arguiles/build/built/
16:59.27HeadachesAboundnot that job.  the other one.
16:59.27brettnemI don't know if it's a supura inside of it.
16:59.41Hmmhesaysthey let you IRC while stripping?
16:59.44brettnemit's an ata, but I didn't think it was the sipura stack
16:59.45sivanahehe
16:59.46HeadachesAboundand besides, how would they know if i was ircing while driving?  the camera can't see the bus.
16:59.52*** join/#asterisk dersteer (n=travis@24-236-197-212.dhcp.aldl.mi.charter.com)
16:59.57HeadachesAbounder, the bus driver.
17:00.02fileGEEKS!
17:00.16HeadachesAboundand well, i have to have something to do while hanging upside down on this pole.
17:00.18arguilebrettnem: I was talking with someone at Cisco who thought it was, that's the extent of my knowledge there
17:01.38HeadachesAboundHurricane Rita might just make things miserable for alot of us.  If it gets strong enough, it may even still be a tropical storm by the time it reaches me.
17:02.09bumblefsckarguile: as soon as you can, upgrade to non-linksys firmware
17:02.25fiber0ptiAnyone that has experience with Polycom 500 series phones; can the lcd screen be used to display who is on specified lines?
17:02.46KattyHmmhesays: neat.
17:02.51Kattyfile: stop that.
17:03.10filemeep
17:03.29filebbs
17:03.38PoWeRKiLLsomeone use te410p ?
17:03.48shido6yes
17:04.22PoWeRKiLLshido66 you do ?
17:04.29shido6yes
17:05.05PoWeRKiLLI got a problem when making ZAP call even if it's busy or unanswered it's written in cdr as answered
17:05.35PoWeRKiLLbecause the zap channel transmit audio to the sip device instead of send ringing or busy sip message
17:06.44tgrmanthe linksys ATAs do run the Sipura stack
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17:19.06fileI'M CLEAN!
17:19.15filescary eh?
17:20.19*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
17:20.21enderSo, I want to give users a prompt when they call my desk and I'm not there, a prompt to call my cell (during work hours).  I'd like them to press a button to call the cell, or stay on the line to leave a message.
17:20.35ManxPowerQ: What is George Bush's opinion on Roe vs. Wade?
17:20.36*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
17:20.44ManxPowerA: He doesn't care how people get out of New Orleans.
17:20.50endermy line isn't in it's own special context, so how would I catch a button press only when people are calling my desk?
17:20.54*** join/#asterisk fiXXXerMet (n=kvirc@ip67-154-236-201.z236-154-67.customer.algx.net)
17:21.14fiXXXerMetHas anyone here used A@H for 40ish users (one location)
17:22.30BruXoasterisk rpm for Mandriva its ok ??
17:24.35*** join/#asterisk bsdee (n=me@c-064-186-237-126.oc1.redwire.net)
17:26.50shido6Powerkill, time to write a billing app
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17:31.37arguilebumblefsck: I fully agree, unfortunately that router isn't even available non-vonage unless you're a licensed reseller or something
17:31.58*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
17:32.09PakiPenguinevening
17:32.50Hmmhesaysi wanna feel the sunshine, shining down on me and you
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17:37.07akrallGuys.. anybody had sound problems with sipuras 84? low volume etc?
17:38.57fiXXXerMetI am confused...  If I have my own broadband service (a T1), than why do I have to use a VoIP provider, if I'll be hosting my own hardware/software and everything?
17:40.19azzie_fiXXXerMet, for the same reason you have two bathrooms in your house ;)
17:40.48endercan somebody help me grok switch => please?  I'm having a bit of difficulty w/ the concept.
17:40.52enderor rather the ordering.
17:41.01*** join/#asterisk Moc_ (n=mochouin@modemcable139.70-131-66.mc.videotron.ca)
17:42.30fiXXXerMetazzie_>  What? lol.
17:42.43fiXXXerMetFor redundancy?
17:43.03azzie_fiXXXerMet, for heck a lot of reasons
17:43.05*** part/#asterisk Senjin (n=g@ip68-11-19-7.no.no.cox.net)
17:43.12enderfiXXXerMet: how will people call you?
17:43.16FuriousGeorgefiXXXerMet: if you have a t1 then you can assign did's to the channels, so i wondered that two
17:43.17FuriousGeorgetoo*
17:43.39enderfiXXXerMet: Somebody has to give you DIDs.
17:44.08fiXXXerMetDIDs?
17:44.57enderfiXXXerMet: Direct Dial.
17:45.04enderfiXXXerMet: phone numbers.
17:45.14enderfiXXXerMet: things people on the outside world cna call to get to you
17:45.26fiXXXerMetOh I see.
17:45.35fiXXXerMetWell they need to stop charging so much :-p
17:45.43*** join/#asterisk perlmonky (n=perlmonk@pix.benchmark-systems.com)
17:45.57*** part/#asterisk perlmonky (n=perlmonk@pix.benchmark-systems.com)
17:46.44enderfiXXXerMet: so find another solution.
17:47.29*** join/#asterisk jero (n=sflphone@savoirfairelinux.net)
17:47.31enderfiXXXerMet: the numbers called have to get to your box somehow.  Over your T1 via Ethernet, through your T1 using dedicated channels for lines, through a partial / full PRI, etc...
17:48.09fiXXXerMetender:  As for another solution...  That's why I'm looking into asterisk :)
17:49.34enderfiXXXerMet: asterisk is a PBX solution.  Again, the numbers have to GET to asterisk in one way or another.
17:50.24enderfiXXXerMet: VOIP isn't bandwidth, it's getting you those numbers over ethernet.  PRI is getting you those numbers via a T1 like line, and a T1 is well, 24 channels of either voice or data.
17:50.30*** join/#asterisk santiago (n=santiago@63.245.86.145)
17:50.52enderfiXXXerMet: some how, some way, callers have to dial your numbers and those numbers have to get from the POTS network over to you.
17:51.41rayvdAnyone here using SER + Asterisk?  I'm trying to figure out if rtpproxy and/or mediaproxy has jitter buffer support (asking on #ser as well)
17:51.47fiXXXerMetender:  Yup, I know.
17:51.59Kattythis company should be beat.
17:52.30*** part/#asterisk santiago (n=santiago@63.245.86.145)
17:55.23PakiPenguinhey , i need to set a variable to value zero after answering how do i do that
17:55.25JerJerDon't know what to do if second ROSE component is of type 0x6
17:55.34JerJerWTF is that?
17:56.06wunderkinyou're pushing up roses instead of daisies :D
17:56.14*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:56.26*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
17:57.55KattyHmmhesays: i want to fight.
17:57.57KattyHmmhesays: get over here.
17:58.00FuriousGeorgeender: regarding a T1 and a DID:  so its really no different than just getting one twisted pair and having the telco lease you a #
17:58.07KattyHmmhesays: bring a helmet.
18:00.57azzie_FuriousGeorge, T1 is sometimes better because it 1. does not have packet loss; 2. does not have compatibility issues 3. does not have DTMF relay problems 4. etc ;))
18:01.07*** join/#asterisk dos000_ (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com)
18:01.49FuriousGeorgeazzie_: i gotcha, i just always thought when you got a t1 you by default got a did, and that for every voip did there was a corresponding channel of a t1 somewhere
18:01.50dos000_anyone know end user devices that support any kind of encription for the rtp stream even xoring !
18:02.24FuriousGeorge~ss7
18:02.30jbotextra, extra, read all about it, ss7 is something that * does not support right now..... given time, who knows :-\
18:02.40Ariel_FuriousGeorge, you can have a PRI with over 100 did's for your office and only have 23 voice channels
18:02.53*** join/#asterisk Cheng29 (n=cheng29@d221-68-210.commercial.cgocable.net)
18:03.04mutilatormine has like..
18:03.04*** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
18:03.06mutilator3000
18:03.12mutilatorand only 1 pri
18:03.13mutilatoro_O
18:03.25Ariel_mutilator, yes you can
18:03.33BrianR___Hmm.. Is there any way to validate a user's voicemail password from the dialplan?
18:03.38mutilatoryes i can what?
18:03.46Ariel_but your more likely to have a busy signal then the one with only 100 numbers
18:03.52klictelor you can have multiple PRI and only on DID
18:04.00azzie_FuriousGeorge, not necessarily
18:04.12mutilatoryeh i know
18:04.18mutilatori only have like 200 subscribers
18:04.18azzie_FuriousGeorge, you can have DID without T1 and you can have T1 without DID :-)
18:04.38mutilatorbut i have like 50 did's for 'every' areacode/prefix in michigan
18:04.41FuriousGeorgeis the exclusice purpose of assigning a did to a data channel voip?
18:05.01Ariel_FuriousGeorge, no it's a price model some use
18:05.39FuriousGeorgeAriel_: i see.
18:05.41Hmmhesayswhy do you want to fight?
18:05.47mutilatorit's fun?
18:05.56mutilatormakes for a less boring day
18:06.10FuriousGeorgeplus, jbot was talking smack about you
18:06.13Ariel_remember the 60th's and 70dy's make love not war.
18:06.14FuriousGeorge~beat Hmmhesays
18:06.17jbotACTION beats Hmmhesays with a large stick.
18:06.29Hmmhesaysamateur night makes for a less boring day
18:07.57KattyHmmhesays: because work is pissing me off.
18:07.59FuriousGeorgei like the heavyweights.  awaiting the klitschko (sp) "lights out Toney" fight
18:08.02*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
18:08.03KattyHmmhesays: they don't /appreciate/ me
18:08.09KattyHmmhesays: now they're awnting me to answer phones for them
18:08.19KattyHmmhesays: and put ringtones and wallpaper on their nextel phones
18:08.30Hmmhesaystell them no
18:08.31Hmmhesayslike I did
18:08.32KattyHmmhesays: next there won't be a computer department AT ALL
18:08.40Hmmhesaysor show up to work an hour late
18:08.42KattyHmmhesays: because there is no marketing at all
18:08.50KattyHmmhesays: even /after/ i made them a brochure
18:08.50Hmmhesaysbecause you were passed out on the floor of your buddies apartment
18:08.52KattyHmmhesays: and flyers
18:09.04KattyHmmhesays: and even did up a legal contract for them to have with new computers sold.
18:09.09KattyHmmhesays: but noooooooo
18:09.16*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
18:09.16FuriousGeorgeKatty: whats the issue with that.  sounds like you got an easy job.  you got bigger plans for the company?
18:09.18KattyHmmhesays: let's put computers on the back burner cause they're not important
18:09.27KattyHmmhesays: we've got More Important things like copiers and nextel phones sell
18:09.29Kattythose bastards.
18:09.38KattyFuriousGeorge: stay out of this.
18:09.43KattyFuriousGeorge: kthx. i'm mad.
18:09.47BrianR___If a user presses '4' to dial out from voicemail, are any channel variables set that would let me know which user dialed out?
18:09.48Uther_Phaha
18:09.53tmccraryhey i have a problem, i have a TDM400 with one FXO module and I have the drivers installed fine:
18:09.54ManxPower~docs
18:09.56jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
18:09.57ManxPower~mailinglist
18:09.58jbotmethinks mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
18:10.02tmccraryChannel 01: FXS Kewlstart (Default) (Slaves: 01)
18:10.02tmccrary1 channels configured.
18:10.10HmmhesaysKatty: i know what you mean
18:10.11FuriousGeorgeKatty: i didnt mean to sound sarcastic.  i was seriously wondering if you had some projects you couldnt get funding for
18:10.14tmccraryHowever, asterisk still says the channel is not found
18:10.15tmccrarychan_zap.c:778 zt_open: Unable to specify channel 1: No such device
18:10.21KattyFuriousGeorge: number one, this is a rant moment.
18:10.21tmccrary[channels]
18:10.21Hmmhesaysi had to be the bitch and fix the owners computer the other computer
18:10.21tmccrarylanguage=en
18:10.22tmccrarycontext=incoming
18:10.22tmccrarysignalling=fxs_ks
18:10.22tmccrarychannel => 1
18:10.25HmmhesaysI was furious
18:10.27KattyFuriousGeorge: and number two, i am a female.
18:10.28Uther_Pdon't take any guff from these fucking swine
18:10.29tmccrarythat's my zapata, i don't know whats going on with asterisk
18:10.34KattyFuriousGeorge: as a side note, do not interrupt a female when she is ranting.
18:10.36tmccraryRegistered tone zone 0 (United States / North America)
18:10.36tmccraryI get that in dmesg, which I assume means the card gets a dial tone
18:10.54ManxPowertmccrary, the output of ztcfg only shows you what is configured, not what is true
18:10.55Uther_Poooh
18:11.08tmccraryoh, is there a way to see what is true?
18:11.11ManxPowermost of the time the fxo module is on port 4.  check the physical card
18:11.16KattyHmmhesays: what pisses me off more than anything is that people around here don't think i work
18:11.20FuriousGeorgewow.  i like my gf a lttile bit more, now :)
18:11.26KattyHmmhesays: they're trying to pawn these stupid jobs off on me.
18:11.36KattyHmmhesays: rather than working on servers i've been downgraded to phone answering.
18:11.39tmccraryon the card, I plugged the cable into the port that had a light, is that correct?
18:11.51KattyHmmhesays: and no one bothered to ask me if i had time to answer phones...which would have been the polite thing to do
18:11.53klicteltmccrary: you did a ztcfg?
18:12.03tmccraryChannel map:
18:12.03tmccraryChannel 01: FXS Kewlstart (Default) (Slaves: 01)
18:12.03tmccrary1 channels configured.
18:12.05KattyHmmhesays: let's just a assume Katty is just sitting around and napping all day!
18:12.10ManxPowertmccrary, look at the physical card.
18:12.12tmccrarythats ztcfg -vvvv
18:12.14KattyHmmhesays: because the IT department obviously runs itself!
18:12.18ManxPowerthe pcb will be labeled
18:12.24HmmhesaysKatty: of course
18:12.33tmccraryif I run ztcfg without arguments, it replies with nothing
18:12.34tmccraryoh ok
18:12.44Hmmhesaysfortunately I take days off here and the people realize just how much gets done back here
18:12.48Hmmhesayswhen i'm not here
18:12.56Uther_Pyou're the one that used you being a woman as an example of something earlier... but it sucks when it works against you instead of to your advantage
18:12.59tmccrarythank everyone
18:13.02tmccraryerr, thanks everyone
18:13.22KattyHmmhesays: wish i could do that. currently dont have any ETO though...
18:13.29Hmmhesaysi see
18:13.40KattyHmmhesays: and now they want me to get my mcse
18:13.42FuriousGeorgeHmmhesays: its rough when you do something not tangible that the people who pay you dont understand.  my bosses always tell me how simple life was before computers
18:13.45Hmmhesayswell if they say i can't, i do it anyway
18:13.47KattyHmmhesays: apparently i'll just do that at home in my free time
18:14.45tmccraryit was on channel 4, you guys rock
18:15.34tmccraryI'm officially an asterisk newb
18:16.36BrianR___Is there an easy way to dump the contents of all channel variables for debugging purposes?
18:16.57Hmmhesaysugh got a 20 t1 unit here losing framing on 4 t's
18:18.01*** join/#asterisk scopeuk (n=Scottywo@cpc3-mfld2-3-1-cust176.nott.cable.ntl.com)
18:19.53*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
18:21.14BrianR___In particular, I'm trying to find out if the "4" outdial feature in asterisk voicemail sets any variables that might let me later identify which user used it.
18:21.14cpatryBrianR___: show application dumpchan
18:21.26BrianR___no dumpchan in 1.0.x?
18:21.39cpatryha, just use it in stable ?
18:22.49BrianR___maybe I should break out the source..
18:22.51tmccrarywhy is it that for an FXO you use fxs_ks and viceversa for an FXS?
18:23.25mutilatorummm
18:23.59Ariel_tmccrary, you signaling is that way (you plug in a pin set to a whole set don't you) Can't plug pin to pins?
18:24.43Ariel_whole/hole
18:25.23tmccraryit works, it works!
18:26.55HmmhesaysKatty: i work about 14 hours a day
18:27.14KattyHmmhesays: i'd find another job if i were you.
18:27.22*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:27.30KattyHmmhesays: it's not like you're not valuable.
18:27.32HmmhesaysKatty: I work for 3 companies currently
18:27.35Kattyk
18:27.37Hmmhesays2 contracts and 1 day job
18:28.11Hmmhesaysthe rest of my time I spend at the bar, or learning new tunes for the band now
18:28.16Hmmhesays;D
18:28.39Kattyor spiking your hair.
18:28.48Hmmhesaysthat only takes about a minute
18:29.16Kattyi can style my hair in less time than you
18:29.37Hmmhesayshaha
18:30.08MocAnyone know how to reach sivana
18:30.24Kattyi really want to do something tonight
18:30.34HmmhesaysI'm going out for 50 cent taps night
18:30.34Kattygoing stir crazy
18:30.38*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
18:30.43Hmmhesays8-midnight
18:30.52Hmmhesaysthen i'm going to stumble into someones yard
18:30.55Hmmhesaysand pass out
18:31.09*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
18:31.09*** mode/#asterisk [+o twisted[asteria]] by ChanServ
18:33.57BrianR___Hmm.. Looks like I can hack around my problem by reading the voicemail login myself and storing it in a variable...
18:34.56Ariel_BrianR___, what are you trying to do with voicemail?
18:36.11KattyHmmhesays: sounds very blah
18:36.25trimi`Does any1 know IAX providers with unlimited calling plans ???
18:36.25*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
18:36.26trimi`Does any1 know IAX providers with unlimited calling plans ???
18:36.32twisted[asteria]bow wow wow yippie yo yippie yay
18:36.44twisted[asteria]twist doggy dogg is in the hizzzouse
18:37.14NORANDOMSnipple clamps
18:37.31twisted[asteria]NORANDOMS, that sure was random
18:37.43NORANDOMSbut sexy
18:37.48twisted[asteria]uhh
18:37.59twisted[asteria]kthxbye
18:38.07NORANDOMShaha
18:38.24trimi`Does any1 know IAX providers with unlimited calling plans ???
18:38.26NORANDOMSi ate kfc buffet for lunch it was good
18:38.38[ViRii]has anyone been successful getting the phones to work via intercom, or auto answer from a specific caller
18:38.48twisted[asteria]i ate japaneese
18:38.53twisted[asteria](not the people, but the food)
18:38.56NORANDOMSwhy
18:39.22Juggie[ViRii], you need a phone that supports setting a perticular extension to auto-answer
18:40.00Juggiei believe the cisco 7960 does
18:40.06[ViRii]polycom for example?
18:40.08Uther_PSushi rocks
18:40.12tmccrarySushi does rock
18:40.19Uther_Punagi
18:40.24Uther_PUNAGI!
18:40.25Uther_P:D
18:40.26tmccraryUnagi is awesome!
18:40.35tmccraryI like my unagi with lots of wasabi
18:40.41tmccrarysounds like an 80's song or something
18:40.53DaminI prefer Futomaki!
18:40.59Uther_PI like my Sushi with lots of Saki
18:41.03DaminFutomaki Haiku is awsome!
18:41.14Juggieyes, [ViRii], i can confirm that the 7960 has a Autoanswer/Intercom mode.
18:42.02trimi`Does any1 know IAX providers with unlimited calling plans ???
18:42.09tmccraryokay, i have setup an fxo that lets you call an asterisk voip phone from a pstn. In order for a voip phone to call phones on the pstn, do I need an FXS?
18:42.11*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
18:42.27BrianR___Ariel_: I need to know (through a variable) which user was logged in when '4' was pressed to make an outside call.
18:42.44trimi`<tmccrary> yes
18:42.55tmccrarythought so, thanks!
18:42.56trimi`or use a softphone
18:43.08tmccrarywhat do you mean?
18:43.12trimi`but you'll have 2 use headsets
18:43.24Kattytwisted[asteria]: you should fight me.
18:43.25Ariel_BrianR___, what does that have to do with voicemail
18:43.27trimi`get a softphone and you can talk using your soundcard
18:43.32trimi`with headset
18:43.36tmccraryoh
18:43.44*** join/#asterisk rob314 (n=rob314@207.58.194.2)
18:43.44Ariel_trimi look in the wiki to see
18:43.49Ariel_~docs
18:43.53jbotfrom memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
18:43.58trimi`<Ariel_> look what ?
18:44.06Ariel_foryour iax provider
18:44.10trimi`i tried
18:44.19BrianR___Ariel_: In voicemail option 3 advanced options, under option 4, a user can place an outgoing call.
18:44.20trimi`no providers with unlimited plans
18:44.30Ariel_check asterlink, teliax voicepulse
18:44.38trimi`i did
18:44.48trimi`asterlink terminates US only i think
18:44.54trimi`have u tried asterlink ?
18:45.01BrianR___Ariel_: So by doing Read(VOICEMAIL_LOGIN) and then VoiceMailMain(${VOICEMAILLOGIN}) I've got a variable I can look into.
18:45.28trimi`<PROTECTED>
18:45.40twisted[asteria]Katty, fight you?
18:45.45twisted[asteria]Katty, how so?
18:45.46Kattytwisted[asteria]: yes.
18:45.54Kattytwisted[asteria]: with a helmet
18:45.57Ariel_trimi`, I don't use any unlimite providers since it's less for me to pay by the minute
18:45.59twisted[asteria]Katty, huh?
18:46.03Kattytwisted[asteria]: i'm pissed off.
18:46.05twisted[asteria]Katty, why would I want to fight you?
18:46.07*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
18:46.10Kattytwisted[asteria]: to relieve tension.
18:46.23twisted[asteria]:(
18:46.29Ariel_BrianR___, does this not take you to a context before making the outbound dialing?
18:46.34Kattytwisted[asteria]: come on, you can take it.
18:46.35PakiPenguinhello , there ,
18:46.40tmccrarywait a minute, it looks like I only need an FXS to connect a Asterisk to a traditional phone, correct?
18:46.43Kattytwisted[asteria]: it's not like you're Hmmhesays's size.
18:46.45twisted[asteria]Katty, yeah, but I can't say i'd fight back
18:46.54twisted[asteria]ROFL
18:46.54Katty:<
18:46.55Uther_Pheh, you shouldn't fight an op... they don't have to punch, they can /kick  :P
18:46.59Hmmhesayssay what?
18:47.00twisted[asteria]nice crack on Hmm-home
18:47.01*** join/#asterisk phpboy (i=flipside@tbnb-165-200-45.telkomadsl.co.za)
18:47.01akrallGuys.. anybody had sound problems with sipuras 84? low volume etc?
18:47.03twisted[asteria]er Hmmhesays
18:47.08BrianR___Ariel_: Yes, it does. But that context is the same for every user.
18:47.10*** join/#asterisk __Elf (n=Elf@elf-laptop.glassfish.net)
18:47.11PakiPenguinmy call is being forwarded to an agi and its getting this 'agi_dnid' => '442070994869' , from somwhere , how do i make agi_dnid as zero , before sending it to the agi
18:47.11[ViRii]how would i make the music play somewhat louder than it currently does?
18:47.14Hmmhesaysi missed
18:47.17KattyHmmhesays: you're a scrawny little thing.
18:47.18BrianR___Ariel_: I need to know which voicemail account was logged in when 4 was pressed.
18:47.19twisted[asteria]Katty, i don't hit women
18:47.20KattyHmmhesays: i might hurt you.
18:47.23Hmmhesaysspry
18:47.25Kattytwisted[asteria]: what if i hit first?
18:47.35twisted[asteria]Katty, usually that's the point I restrain
18:47.40twisted[asteria]not hit back
18:47.42HmmhesaysKatty doubtful
18:47.43Kattytwisted[asteria]: restraining would just cause more tension
18:47.49Duttshey guys anyone know anywhere on the net where I can get some free prompts for my * box? pretty much after some voicemail ones? At the very least just the one facing incomgn callers, the 'the person you are trying to reach is not in, pleas eleave a message after the tone type'
18:47.49Kattytwisted[asteria]: hold a mat or something.
18:47.54twisted[asteria]i could do that
18:48.12[ViRii]how do i amplify the volume of the music when someones on hold.? just isnt loud enuff
18:48.22Uther_Pwhat if you hit him... and he started laughing?
18:48.26twisted[asteria]i've helped in kickboxing and stuff by holding mats and the gloves and whatnot
18:48.39Uther_Pthat used to piss my exwife off big time
18:48.46Kattytwisted[asteria]: i know! get SwK!
18:48.47twisted[asteria]hahaha
18:48.50Kattytwisted[asteria]: he'll fight me!
18:48.58Kattytwisted[asteria]: and probably hurt me in the process, heh.
18:49.04twisted[asteria]Katty, no that'd be bad
18:49.50Kattytwisted[asteria]: eh, i'm actually a little better now.
18:49.56Kattytwisted[asteria]: work keeps pissing me off
18:50.33Uther_PI can see the headlines:  "Crazy woman kills IRC Operator after he wouldn't defend himself..... 'his honor and chivilry just made me angrier!' "
18:50.58KattyUther_P: i wouldn't hurt twisted[asteria]
18:51.01KattyUther_P: he's too nice.
18:51.15tzangerhttp://www.jwz.org/images/hamster.jpg
18:51.26Corydon-wKatty: oh, but you never know... he might like that sort of thing...
18:51.37Uther_Pa woman scorned can do lots of things they said they would never do
18:51.57KattyUther_P: probably. i have no doubt of that. but this one has androphobia, so it's HIGHLY unlikely.
18:52.19Uther_Pwhat is androphobia
18:52.27Kattygoogle (=
18:52.31Ariel_BrianR___, you set the context for them to dial out dialout=fromvm  this context you can get info from your use before allowing him to dial out.
18:52.34Corydon-wFear of men
18:53.13Corydon-wFrequently associated with rape victims
18:53.31tmccraryhaha! I don't need an FXS to make calls from my voip phone to the regular PBX
18:53.39twisted[asteria]hey wait
18:53.45twisted[asteria]er
18:53.47twisted[asteria]wrong window
18:54.25tzangerandrophobia: the fear that your call will come into the wrong context
18:54.31*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
18:54.33akrallGuys.. anybody had sound problems with sipuras 84? low volume etc?
18:55.01Uther_Pbwahahah
18:55.12Uther_Phttp://www.jwz.org/images/hamster.jpg  <-- thats farkin helarious
18:55.30Uther_P"if your hamster looks like this, you did something wrong, something very very wrong"
18:56.27tzanger:-)
18:57.06Uther_Pgotta love that picture of the hamster with a meat cleaver taped to it
18:57.54tzangeryes
18:58.02tzanger"your knife is not well matched to your hamster"
18:58.06NORANDOMShaha
18:59.05Uther_Pdoes anyone here know if a 250MB zip drive can read 100MB zip disks?
18:59.32akrallanybody using unicall with r2mfc?
18:59.45NORANDOMSUther_P: it can
18:59.50Uther_Pcool, thanks
19:00.41*** join/#asterisk mut (n=animenod@65.111.201.79)
19:00.44NORANDOMSanyone use postfix with virtual domains?
19:00.54mutfsckin non regged nick channel bs
19:01.10LoRezNORANDOMS: someone in #postfix might have
19:01.13tzangermut: amen
19:03.30*** join/#asterisk clinthome (n=clinthom@24.75.94.170)
19:03.55Ariel_Uther_P, yes it can
19:05.38akrallanybody using unicall with r2mfc?
19:05.47mutilatorhttp://www.cnn.com/2005/WORLD/americas/09/21/fake.emergency.ap/index.html
19:05.49mutilatorclassic!
19:07.13Uther_Pheheh
19:07.41Uther_Pbut he's a president.. he should be allowed such discretions as emergency soccer landings
19:07.51*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
19:09.08*** part/#asterisk ThomasBakketun (n=thomas@i170227.dsl.tjukkband.no)
19:12.30EssobiAny SER experts here?  I can't seem to get the contact field changes on a 200 reply.
19:13.11*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
19:13.21phpboyI'm having trouble receiving calls from my pstn...
19:13.24phpboyexten => s,1,Dial(SIP/10,20,tr)
19:13.38phpboythat's what I've got and according to my readings.. that's correct
19:13.51Ariel_what does it do and why do you have the r
19:14.21EssobiIf you need to ask why he has the r...
19:14.23Essobi:)
19:14.45Ariel_r should not be used unless your doing r2 or have something broken
19:14.48phpboywell, I'm dialing my ISDN number from the outside(pstn) and I want it to ring my sip extention 10.
19:14.54phpboywell
19:14.56ManxPowerphpboy, you have an analog line?
19:15.03phpboyno, ISDN
19:15.09phpboydialing my ISDN card
19:15.22ManxPowerexten "s" is only used when the destination number is not known i.e. analog
19:15.30ManxPowerlook at the console, you should see a rejection message
19:15.39phpboywell, that's the thing
19:15.48phpboyI'm running my consol... asterisk -vvvvvvvvvvr
19:15.50phpboyno msgs
19:15.54phpboyeven sip debug
19:16.03phpboyI dunno how to activate the debugger for that :/
19:16.24ManxPowerphpboy, you are not using some silly preconfigured asterisk, are you?
19:16.56phpboynope
19:16.59phpboyraw install
19:17.04phpboythis is what I've done so far...
19:17.10ManxPowerthen rename /etc/asterisk/logger.conf to /etc/asterisk/logger.conf-disabled
19:17.12ManxPowerstop asterisk
19:17.16ManxPowerstart it as asterisk -cvvv
19:17.19ManxPowerthen try a call
19:17.46ManxPowerif you don't see the call come into the console, then asterisk is not seeing the call and you need to look at your configs for the isdn dfriver/card
19:17.47phpboyinstalled asterisk and bristuff... configured my kernel to get the ISDN card going
19:17.55*** join/#asterisk adrianv (n=yoyo@193.239.135.5)
19:17.57phpboyso I can dial calls
19:18.09phpboyand it works, so I know the ISDN card is correctly installed and configure
19:18.12phpboyconfigured
19:18.33ManxPowerphpboy, you do not know that.  You know it's correctly configured for OUTGOING calls.
19:18.55adrianvhi all
19:18.58*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
19:19.02*** join/#asterisk hellagony (n=egutierr@200.121.129.180)
19:19.05phpboyok, ok
19:19.19phpboylemme do as you say
19:21.25Uther_Phttp://www2.cytecsys.com/~uther/macros.conf  <-- what do you think of my new macro for connecting extensions
19:21.54*** join/#asterisk zedkatuf (n=knoppix@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
19:22.27phpboyhmmm
19:22.34phpboyI'm getting nothing on the colsole
19:22.34phpboy:/
19:22.54blitzrageahoi all
19:23.11Uther_Pa-hoy hoy
19:23.30JerJerok people iaxcomm does not work - do not use it
19:23.35blitzragehaha
19:24.01akrallanybody using unicall with r2mfc?
19:24.16JerJeri can tell anytime someone uses iaxcomm via our shit - we get flooded with received mini frame before first voice frame
19:24.18JerJermessages
19:24.19ManxPowerphpboy, stop asterisk and then start it as asterisk -cvvvddd
19:24.48JerJerthen 2 minutes later a support ticket shows up "one-way audio"
19:24.52blitzrageok, I have a question (javascript) - and I'm going to ask it here even though its off topic :)     I have cmdname = eval("document.test.Button" + conbtn + "_Command.value");. In IE if the form variable I'm trying to evaluate doesn't exist, I get an error "document.test.Button1234_Command" is null or not an object. Is there a way I can "catch" that and doing something? I don't want the error to be reported it to the brow
19:25.00blitzrageI want to actually do something with it...
19:25.33*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
19:25.38blitzragewhen I do the check, the hidden form variable may, or may not, exist
19:26.14phpboynope
19:26.16phpboystill nothing :<
19:26.18*** join/#asterisk asdasda (n=cems@85.101.15.243)
19:27.07*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
19:27.31ManxPowerphpboy, then the call isn't getting to Asterisk.
19:27.36ManxPower~mailinglist
19:27.37jbotsomebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
19:27.41ManxPowercheck the mailinglist archives
19:28.12Ariel_phpboy, do you have the right context setup in the ISDN card setup?
19:30.13ManxPowerAriel_, it should still show a console message
19:30.53blitzragecool - the Asterisk book is number 1 in the telecom category on Amazon!
19:31.09Ariel_blitzrage, is it shipping yet?
19:31.34blitzrageAriel_: yeppers - shipped from O'Reilly to book stores last Monday
19:32.29blitzrageAriel_: and number #440 rated overall!  We were 1237 2 days ago, 580 yesterday
19:32.40*** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net)
19:32.41blitzragelook out Harry Potter! :)
19:32.50Ariel_26.97 is a good price as well
19:32.51Uther_Phaha, thats awesome
19:33.02Uther_Pwhat animal did they use
19:33.05Ariel_Great job....
19:33.32jdv79shido6, hallo
19:33.35Ariel_sorry 26.37 strange price.
19:33.42shido6hello
19:33.57shido6scared the jesus out of me
19:34.07tmccraryasterisk is so awesome
19:34.13shido6yes it is
19:34.13shido6:)
19:34.19blitzrageUther_P: starfish
19:34.21ManipuraI have an IAX hardphone... Is there any service someone offers where I can get it hooked up to a PBX so I can get a DID sent to it? I don't quite have the money for a server yet
19:34.31Uther_Pa starfish... why did they choose a starfish
19:34.33shido6Manipura, NuFone? :)
19:34.43*** join/#asterisk clive- (n=pirch@ndn-165-151-146.telkomadsl.co.za)
19:34.59blitzrageUther_P: because we didn't want the bird they originally gave us :)
19:35.04Uther_Phaha
19:35.16Uther_Pyou gave them back the bird? :P
19:35.37clive-has anyone seen the error message "max trunk data space exceeded"  with iax calls, and then the call just gets dropped
19:35.58Uther_PI'll have to get my company to buy a copy of this book
19:36.27blitzrageUther_P: /msg jbot thebook
19:36.47ManxPowerI'm currently in a town of 6,000 people in Texas.  So I go down to the local Radio Shack (which is located inside the local Ace Hardware store) looking for an ethernet switch.  They said they don't carry them, but the local Feed and Cattle Supply store did.  So we went down to the local Feed and Cattle Supply store and sure enough, they had a small computer repair and sales department.  They were out of stock, but could order a
19:36.48ManxPower<PROTECTED>
19:37.22ManxPowerI expected to see a mad hatter and a cat.
19:37.28Uther_Pdamn, its $40 on o'reilly's site, and only 27 on amazon
19:37.32shido6LOL
19:38.11*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
19:38.18Uther_Phaha, damn
19:39.16Uther_PManxPower:  what town are you in?
19:39.34ManxPowerUther_P, Atlanta TX, about 30 miles from Texarkana TX
19:39.54Uther_Pah
19:40.09Uther_PI'm in dallas... but texas has thousands of towns like what you described
19:41.58ManxPowerAt this point the the top of my list of perm places I might be is Auburn AL
19:42.04phpboyManxPower: I got asterisk to see the call
19:42.06phpboyI put
19:42.09phpboyincomingmsn=*
19:42.10clive-~seen stevek
19:42.20jbotstevek <n=chatzill@64.251.119.55> was last seen on IRC in channel #asterisk, 2d 1h 37m 39s ago, saying: 'arrivederche'.
19:42.21phpboyin my modem.conf
19:42.30ManxPowerphpboy, I'm in the USA, we don't use BRI here.
19:42.36phpboyhere's the werid thing though... if I put the real MSN number there... it doesn't 'see' it anymore :/
19:42.40clive-phpboy howzit, what you trying to do?
19:42.43ManxPoweronce the call comes in I can help.
19:42.49phpboyu guys are lucky
19:42.59phpboyManxPower: I got it to 'see' the call
19:43.02sigwerkmorning people
19:43.04phpboyshall I paste you the msg?
19:43.07phpboyclint_: u in SA?
19:43.11ManxPowerphpboy, only if it's 1 or 2 lines.
19:43.26blitzrageUther_P: yah - that'll be so they don't compete with their resellers
19:43.30phpboypastebin?
19:43.44ManxPowerphpboy, if it's more than 2 lines, use pastebin
19:43.47ManxPower~pastebin
19:43.51jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/
19:43.58ManxPowerI'm getting REALLY sick and tired of hurricanes.
19:44.10sigwerkaye
19:44.11*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
19:44.19sigwerkgot a few coming our way too
19:44.19blitzrageManxPower: another cat 4 on its way...
19:44.28*** join/#asterisk buddah (n=djbrianc@67.110.253.129)
19:44.28tzangerManxPower: looks like you need to get used to it... the 20 year lull is over
19:44.31blitzrageManxPower: I don't deal with natural disasters here in Canada :)
19:44.35clive-phpboy, no clint here...:), yes I am in sa
19:44.36phpboyhttp://pastebin.ca/23540
19:44.42ManxPowerblitzrage, I hate you.
19:44.43buddahanyone know the default login/password for linksys PAP2-NAs?
19:44.44sigwerkcat 4 coming to us as well in hawaii :/
19:44.49sigwerkbut hopefully spinning north
19:44.51blitzragetzanger: pfft - no such THING as global warming
19:45.03phpboythere I dialed... let it ring for a bit and then hung up
19:45.07tzangerblitzrage: uh, it's a CYCLE
19:45.12tzangerjust like the 11 year solar cycle
19:45.35clive-phpboy what hardware are you using
19:45.44phpboyDuxbury ISDN card...
19:45.46Ariel_ManxPower, your getting sick of them I am already sick of them.. There a real pain in my side.
19:45.47blitzragetzanger: hrmmm.... I don't think I believe that - warmer gulf waters would also contribute to more frequent and stronger hurricanes
19:45.54phpboyManxPower: what you think of that?
19:45.55*** join/#asterisk zotz (n=zotz@24.231.36.100)
19:46.03ManxPowerphpboy, no idea
19:46.15ManxPowerblitzrage, marry me and I'll move to Canada
19:46.16tzangerblitzrage: yes but it's also understood that the hurricanes are in a cycle of their own and that we've had a relatively quiet last couple of decades
19:46.27phpboyshit :/
19:46.32clint_Has anyone experienced PRI signalling issues w/ Digium vs Dialogic?
19:47.11blitzrageManxPower: sorry - I'm not into marrying anyone :)
19:47.29Uther_Pdialogic, heh.. ibm still makes those?
19:47.30ManxPower<PROTECTED>
19:47.38Uther_Perr, was it ibm or intel that bought dialogic
19:47.52ManxPowerblitzrage, you may not get Storms of Doom, but you get Blizzards of Doom
19:48.00Uther_PI used to have a voicebrick, that think sucked ass
19:48.02tzangeryes but blizzards of doom are FAR easier to handle
19:48.13tzangerkeep enough food and blankets in your residence for a week and you're golden
19:48.16blitzragetzanger: agreed - I just stay inside anyways :)
19:48.29blitzrage<----- hating on JavaScript
19:48.52*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
19:49.18EssobiDon't hate the language, hate the syntax. :)
19:49.25azzie_hurricanes sucks! My lawyer is in Florida and I can't get ahold of him
19:49.31azzie_oh
19:49.32Uther_Pthe syntax is the language
19:49.43Uther_Pmostly
19:49.44EssobiI was being asinine, you turd.
19:50.04Uther_Phah, you got the ass part right :D
19:50.19Essobihah
19:50.25funxionazzie_ where n florida Im in miami everything is business as usual dot let him use the hurricane as an excuse unless hes in the keys
19:51.30azzie_funxion, oh
19:51.53azzie_do they teach to lie in law schools?
19:52.00scopeukhas any one tried gettign caler id back list into a web page?
19:52.12funxionand if he is in the keys and hes anything other than a dui lawyer you should fina new one
19:52.27JerJerwhy do you think they are called Liars
19:52.30JerJerer lawyers
19:52.31Ariel_I am in miami and it's just fine here.
19:52.44funxionAriel_ where
19:53.00Ariel_West Kendall by the Tamiami Airport
19:53.04funxiontru
19:53.11funxionI live in Pinecrest
19:53.27Ariel_all the rich peoples area...
19:53.33funxionlol
19:53.55funxionhey Im living with mysister cuase I lost myh house in the divorce
19:54.03drrayis she hot?
19:54.06Ariel_funjon, sorry
19:54.09tzangerfunxion: that sucks, sorry
19:54.32Ariel_funny thing is my address is Miami and I am actually 16 miles outside of miami
19:54.57funxionno need to be sorry there was no kids involved
19:55.05funxionI can start over
19:55.16*** join/#asterisk scfrec (i=scfrec@scfrec.compic.ee)
19:55.22funxion16 miles
19:55.26funxionu live in hojmestead
19:55.28scfrechello. h323 make segmentation fault after: "     -- Executing Dial("SIP/scfrec-1bee", "H323/000#0000+37260000008@194.xxx.xxx.xxx") in new stack"
19:55.28tzangerfunxion: that's good then... there were with mine and it's not been too bad but then again my ex isn't a ho-bag "take him for all he's worth" type either
19:55.42Ariel_funxion, no but Krome is just around the conner
19:55.52funxionryte on
19:56.05funxionend of kendall huh
19:56.12funxionthats where I used to live
19:56.52funxionnothing like a shake @ grandma's garden
19:57.04Ariel_just sucks that it takes an hour to drive into miami from here
19:57.18funxiontzanger I've found that most women in miami are the hobag take him for all hes worth type
19:57.32tzangerfunxion: I'm in Canada :-)
19:57.36*** part/#asterisk scopeuk (n=Scottywo@cpc3-mfld2-3-1-cust176.nott.cable.ntl.com)
19:57.39drrayfunxion - you need the 8 time rule
19:57.55funxionI want to move to Canada
19:57.56Ariel_tzanger, we get in the winter allot of them from Canada here
19:57.58tzangerwe just get along better apart is what it comes down to, and we're very much together with respect to the kids...  I dunno I describe it as "mostly amicable"
19:58.09tzangerAriel_: yep I bet... all the snowbirds as we call 'em
19:58.54tzanger??
19:59.51Ariel_tzanger, we get allot of the French Canadian ones in Miami Beach around Dec/Jan
19:59.54funxionyeah I was gonna say most of the canadians we get down here are the speedo wearin french canadian
20:00.09funxionand they're rude as hell
20:00.20Ariel_rofl yes they are
20:00.38funxionnor do they know how to drive
20:01.55Ariel_funxion, I don't blame them for that. I blame north around NW 103rd street people for that.
20:02.16funxionlol
20:02.51sylelol
20:02.58sylethose are teh canadians from quebec
20:03.05syleunlike the rest of canada
20:03.06BrianR___yay. I got the user-programmable outbound transfer to work on asterisk voicemail
20:03.10Cheng29LOL at the comment about french canadians
20:03.29funxionI want to move to vancouver
20:03.48Ariel_funxion, yes that is a nice place. or Victoria Island
20:03.50funxionI hear the women are hot there
20:04.03Ariel_Just need a good rain coat
20:04.06syleyes
20:04.08funxionalthough miami's got the hottest around
20:04.20funxionlove them latinas
20:04.25sylebut all women are very money hungry in vancouver and toronto
20:04.29sylekeep that in mind hehe
20:04.39funxionno worse than miami
20:04.44tzangerAriel_: you poor bugger
20:04.48ManxPowerI don't suppose anyone knows anyone that works for SBC?
20:04.58Ariel_I don't have to worrie about that I have a great wife already
20:05.06syleyou;ll do well with most of their parents if you have a good job and good investments
20:05.45*** join/#asterisk sudhir492 (n=sudhir@12.109.60.108)
20:05.53sudhir492hi all
20:06.10sudhir492anyone here using Polycom phones
20:06.12sylethe rest of canada noone cares, as long as you treat their daughter right they are happy hehe
20:06.44funxionryte on
20:06.48Ariel_Argh 155mph Rita cat 5
20:06.51funxionsyle where do you livce
20:06.53funxionlive
20:06.57sylewinnipeg
20:06.58tzangercat5 now?  yikes
20:07.10Ariel_sudhir492, yes great phones
20:07.12sudhir492I can always call out from my polycom phone but cannot receive calls from time to time
20:07.17sylebut i have lived on vancouver island, vancouver, toronto, calgary, phoenix etc
20:07.38drrayI'd love to live in vancouver BC
20:07.44Ariel_syle, wow from wet to dry
20:07.56syleyeah helps my asthma
20:08.19sudhir492Ariel_: Yes, great phones, but do not know what is the way out of this? I am using 2.6.1 sip 1.4.1
20:08.22sylebut that is not the real reason i moved to winnipeg
20:08.24syle2 reasons
20:08.34syle1) family 2) cheaper real estate
20:08.51Ariel_sudhir492, update to 1.5.2 or .3 great sip setup.
20:09.01sudhir492and on another, 2.6.2, sip 1.5.2
20:09.02funxionwhats a house in vancouver gpo for?
20:09.06Ariel_and use the ftp setups to get them up and running
20:09.07tzangerwhat's wrong katty
20:09.12tzangerfunxion: WAY too expensive in vancouver
20:09.12syleat least 200k
20:09.14*** join/#asterisk patrick^ (n=patrick_@pc-0-34.mountaincable.net)
20:09.15tzangerhell alberta's expensive
20:09.28Kattytzanger: it would take 2 hours to explain. i don't have the energy.
20:09.31sylea good house you will be happy with will run about 300k
20:09.42sylein winnipeg they run about 130k up
20:09.47syleCDN
20:09.52sudhir492Ariel_: how do you setup your phones?
20:10.14funxionmy house in miami is going for over 800k
20:10.24funxionnot that bought it for that
20:10.37mutilatorwatch out for tornados!
20:10.46syleprices in Toronto and Vancouver are comparable to all US prices
20:10.53tzangeryes
20:11.04tzangermaybe sell in the spring
20:11.07syleif you don;t mind the cold winters you can get cheaper real estate
20:11.17Ariel_sudhir492, via the ftp setups and they work great I only configure one line on them.
20:11.18tzangermove closer to my kids, maybe rent a house
20:11.19sudhir492I am already living in a trailer park. Cannot afford ridiculous house prices in Northern VA
20:11.21tzangersince an apartment's small
20:11.53sudhir492What line do you configure?
20:11.57*** join/#asterisk spackle (n=spackle@209.234.83.19)
20:12.19syleif i didn;t have kids i;d live in a trailer park or condo
20:12.22sylesave some money
20:12.29tzangeryeah
20:12.36tzangercondo won't save you anything, they're as expensive as bloody houses
20:12.43sylethats true
20:12.47tzangerbut yeah rent might be an option for the next few years
20:12.47syletrailer park is the way to go
20:12.51tzangertrailer park too yeah
20:12.54tzangerI could do that
20:13.33sylecondo fees these days are insane , i remember checking on that a month ago
20:13.41mutilatori'de live in a hut on the beach
20:13.49mutilatorlong as i had wifi from something i'de be set
20:13.53tzangeryup same here
20:13.56blitzragetzanger: people in Alberta are getting a $400 cheque this year because of the oil prices
20:13.57tzangerrent a farmhouse
20:14.05tzangernice
20:14.12tzangercan I travel to alberta to pick up my cheque?
20:14.17blitzragetzanger: I wish :)
20:14.28syleso you finally pay your condo off, you still have like 300 a month in condo fees and then their is your property tax, almost makes you want to move to the carribean sometimes
20:14.30blitzragew00t!  jsmith helped me figure out my problem with JS
20:14.41sudhir492syle: you are talking condo fees, my hoa dues in Atalanta used to be $100 per month, just for the stupid pool which I never used :-(
20:14.44Kattyyay for geeks.
20:14.52drraycondo fees cover water/garbage and maintence on the property
20:15.07mrfrenzyokej
20:15.09mrfrenzyoops
20:15.12blitzrageKatty: aye
20:15.13NuggetI pay $100 a month for HOA fees.
20:15.32spacklesyle, there should be a condo for geeks, no pool, solar electricity and extra outlets.  wifi everywhere
20:15.46sylehaha yes, they have that in phoenix
20:15.48tzangerI have no water/sewage fees since I have my own tank and my own well
20:15.51drrayspackle, it would stink
20:15.51sudhir492But at least they collect your trash free, not mine
20:15.57sylesome company down there sells solar
20:16.04spackledrray, good point
20:16.05blitzragetzanger: www.ipm2005.com !!!! :)
20:16.28Kattyblitzrage: !!11oneone
20:16.33tzangerblitzrage: yup
20:16.33*** join/#asterisk epablo (n=epablo@200.75.139.188)
20:16.37epabloHello people
20:16.39tzangerright across the highway from me
20:16.47sylewell you could do that for your own house, if you can afford 30k then you can get the equipement necessary to power a good part of your house on solar
20:16.50blitzrageKatty: hey hey hey!
20:17.06blitzragewind mills!
20:17.19blitzragetzanger: cool - IPM was hulla fun when I went to it in grade 7
20:17.27tzanger:-)  I run our booth tomorrow morning
20:17.27blitzragetzanger: it was just outside of Petrolia
20:17.32syleits worth it over the course of 20 years if you plan on staying in the same house, if not then i;m sure that investment would increase your real estate price anyways
20:17.32blitzragetzanger: haha :)
20:18.00epabloI've got a weird audio problem using SIP.  I can heare voicemail but I can't talk between two ext.  Any ideas where are should look?
20:18.01blitzragetzanger: I remember going around the place looking for people giving out helium balloons - crazy voice :)
20:18.23sylelol
20:18.24tzangerblitzrage: you have a crazy voice already
20:18.31blitzragetzanger: this is true
20:18.32Ariel_epablo, nat issues
20:18.32ManxPowerI don't suppose anyone knows anyone that works for SBC?
20:18.33bkw_woops RITA turned into a whore bitch from hell
20:18.34sylei think everyone has done that in their life at some point
20:18.43blitzragebkw_: LOL - its true
20:18.47Ariel_ManxPower, not me
20:18.51blitzragebkw_: went from cat 1 to cat 4 overnight
20:18.54bkw_ya
20:18.58bkw_its headed for houston
20:18.59tzangerI dunno I bought my place for $175 I think I could sell for a little more than that now
20:19.00bkw_haha
20:19.06bkw_where all the people for LA went
20:19.12ManxPowerbkw_, Yeah.
20:19.30tzangerlive in a rented farmhouse for a few years, let the bubble burst then buy 50ac somewhere cheap
20:19.36epabloAriel_: I think it comes from that. But I have the nat=yes and canreinvite=no on all my exten.
20:19.39*** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk)
20:20.09ManxPowerat least TX doesn't have to worry about the levies breaking
20:20.11Ariel_epablo, are you forwarding ports 10,000 - 20,000 for sound?
20:20.51Ariel_ManxPower, correct but looks like there thinking some storm serg will still hit LA some.
20:21.18ManxPowerAriel_, *nod*  It's totally screwing up plans for my friends to go back to New Orleans to move their stuff out.
20:21.29blitzragetzanger: EXACTLY - buying a big house right now seems like the dumbest idea in the world to me
20:21.33epabloAriel_: ??  I have a broader range defined in my rtp.conf and all of it open in my iptables (I tried with it down with the same result)
20:21.44spackleRita at cat 5?
20:21.47syledepends blitzrage
20:21.48ManxPowerspackle, yup
20:21.58syleif your currently renting, then no its not
20:21.58blitzrageI think the housing market is setup to crash
20:22.04blitzrage<-- renting
20:22.08Ariel_epablo, do you have your externip= and localnet= setup
20:22.09drrayIn seattle we pay $1000 for a one bedroom, I can get a condo for $1200
20:22.21tzangermy mortgage is about that
20:22.23Uther_Panyone else get the feeling that Mother Nature is trying desperatly to correct a major mistake in allowing mankind to make it this far?
20:22.24spackleblitzrage, how do you figure?
20:22.26pauldyblitzrage, it should and it could but it probably wont
20:22.27sylethen i;d do it, cause your interest rates are low on mortgages right now anyways
20:22.29blitzrageI live in a 3 story townhouse for $1650 w/ utils
20:22.33epabloAriel_: The server is not behind nat.  Only the clients
20:22.37ManxPowerI'm very glad I'm a renter.
20:22.43blitzragepauldy: bubbles burst - thats what they do
20:22.47Ariel_Uther_P, no
20:22.51pauldyblitzrage, its being propped up by lots of investment capital and newbie realestate investors
20:22.54ManxPowerMy landlord can deal with the damage of 4ft of water from Katrina
20:22.54drrayManx, when do you go home?
20:22.57Uther_PI don't blame her
20:23.03ManxPowerdrray, I don't go home.
20:23.10pauldyblitzrage, and who knows how long they can keep it afloat
20:23.15sylemanxpower your only ever happy being a renter if you;ve never owned a home, think of all the equity your loosing!
20:23.29tzangerpauldy: yes but for how long... tha'ts what I'm banking on ... it bursting within the next few years
20:23.30drrayManx - move to austin, :)
20:23.35spacklehouse insurance isn;t that outrageous.
20:23.36ManxPowerI could move back into the place I rented in about 60 days, but there's nothing left in the area.
20:23.44blitzragebeing a renter gives me a lot of flexibility - I don't like staying in one place too long
20:23.57ManxPowersyle, and you are only happy owning a home until the roof blows off or you get termites
20:24.04PakiPenguincan someone please help me with agi_dnid
20:24.06blitzragetzanger: yep - I think so too - all these huge suburbia houses will be teh cheep :)
20:24.07pauldyI wouldn't count on it within the next 7-10
20:24.12tzangerManxPower: nah I love owning a house
20:24.13sylethats not an excuse, you could just rent out your place, build more equity, and buy a new one blitz
20:24.15drrayI grew up in Austin, and there are nerd jobs there
20:24.33tzangersyle: yeah but then you're riding the entire real estate bubble and it'll take forever for that house to pay off
20:24.40ManxPowerI LIKE having someone else deal with lawn care, broken pipes, termites, dry rot, roof leaks, etc
20:24.54ManxPowergotta run
20:24.54tzangerheh
20:24.55Ariel_roof leaks I got 2 to fix
20:24.56blitzragesyle: well, hard to do when you're a self employed consultant who just graduated from school with a debt a few months ago :)
20:24.59sylethink is when you are ready to retire your cashing in on hundred of thousands of dollars
20:25.01tzangerI like having the equity in the house
20:25.01spackleI can see why you would like somebody else to handle the maint.  wouldn't mind that.
20:25.09funxionwhat ever happened to the ds3000p?
20:25.13drrayI'm kind of interested in fixing and working in my own place
20:25.16syleowww
20:25.18tzangerproperty almost always goes up in price, it's a very stable investment.  houses are a little more shaky :-)
20:25.18Uther_Pand what if the value completely plummets?  then your equity goes down the crapper
20:25.23Ariel_funxion, still in development
20:25.33tzangerUther_P: which is why I'm trying to hop off this bubble just beofre it blows
20:25.45sylewell when youhave the money i;d definately do it then
20:25.47funxiondamn I finally find an application for it and its not out yet
20:26.07blitzrageok, back to work!
20:26.08niZonhas anyone got audio working using xorcom rapid?
20:26.13Uther_Ptzanger: better jump quick :D  and remember, there is always another bubble under it waiting to pop :P
20:26.17Ariel_we have a very different bubble here. Our keeps going up even when others drop.
20:26.23jdv79work? people still do that?
20:26.28tzangerUther_P: yeah I know, I'm planning on selling in the spring I think
20:26.30funxionAriel_ I agree
20:26.34blitzrageits the only way to pay bills :)
20:26.46blitzragekinda wish I just had a desk job - would get paid even when not being productive :)
20:26.47klictelAriel what bubble are you on?
20:26.47funxionalthough the condo market is already seeing a decrease
20:26.47tzangerI was going to try and rent out the house and live in an apt but this bubble's gonna go soon I think
20:26.48Ariel_blitzrage, where is my bood
20:26.53Ariel_bood/book
20:27.06blitzrageAriel_: its at http://www.oreilly.com/catalog/asterisk :)
20:27.07Ariel_klictel, Miami Real Estate.
20:27.29Ariel_blitzrage, I have it on order aready from Amazon but have not gotten it yet...
20:27.39*** join/#asterisk extremis (n=extremis@cpe-24-175-54-186.houston.res.rr.com)
20:27.49extremisif I don't go with nufone, who should I get access with
20:27.58blitzrageAriel_: ahhh - it just started shipping from the warehouse to bookstores last week - and I've been told sometimes it takes a while - don't worry, I havent' gotten mine yet either
20:27.58Ariel_lots
20:28.00extremisjust something to handle 6 lines while rita destroys houston
20:28.11Ariel_lol
20:28.38funxionthat suxs
20:28.43Uther_Pits a vaccine!
20:28.45Ariel_extremis, asterlink, voicepulse, there are many
20:28.51funxionIm glad rite barely touched us in miami
20:28.51blitzragemixnetworks
20:28.51Uther_Phumans are the real natural disaster!
20:28.52blitzrage:)
20:28.53Uther_P:D
20:28.59blitzrage<-- works for mixnetworks
20:28.59extremisI need a good value
20:29.01sylei wouldn;t be discouraged by buying property just because prices are high, all that is going to happen is when mortgage interest rates go back up, the prices on homes may drop like 30k, time you spent as a renter you could have paid that much already in rent
20:29.05funxionUther_P I agree
20:29.21klictelshit i just read that it's already a cat 5
20:29.29extremisI will be connecting my * box in houston, in Oaklahoma city, and then maybe newyork if it goes on for a while
20:29.42extremiswhat network has low latency from most areas of NA?
20:29.45sylewhere it makes sense to wait is if your buying rental property and have alot of cash to put down
20:30.02funxionis anyo ne else using * over satellite?
20:30.17fileketchup for all!
20:30.56tzangerAriel_: yeah that sucks
20:30.59tzangeryou need private investors
20:31.05Uther_Pheh
20:31.14Ariel_tzanger, yes but don't have any rich uncles
20:31.38tzangerthe company I work for now couldn't get a loan from the bank when they started out... now that they're doing well the banks are tripping over themselves trying to get loans but my boss (the owner) tells them to fuck off, they weren't around when he needed them so they get zero business from him now.
20:31.47extremisis nufone my best bet?
20:31.52tzangerAriel_: me either... there are places to locate local private investors
20:31.56extremisits 2 cents a minute, and they don't have a local houston number
20:32.06*** join/#asterisk Assid (n=assid@203.115.64.60)
20:32.13Uther_Ptzanger:  thats just busniess
20:32.23Uther_Ps/usni/usin/
20:32.28tzangerUther_P: yup, and it's just business to reject them now too
20:32.29Assidheya
20:32.29Ariel_extremis, I use voicepulse connection for my inbound and nufone/voipjet for my outbound
20:32.30tzangerI'd do the same thing
20:32.32Uther_Pheh
20:32.39Uther_Pnot if they have the best rates
20:32.40tzangeryou won't take a chance on me when I need it, why should I use you for anything whatsoever
20:32.41AssidAriel_: i do the same
20:32.50tzangerUther_P: we don't need loans now, we have all the capital we need
20:32.56Assidbut i use voicepulse sometimes too for outgoing
20:33.00Ariel_been saving me money that way.
20:33.03Assidfor atleast 1 of my installations
20:33.12tzangerone of my second cousins had a similar problem when trying to build his house
20:33.24tzangerno bank would give him a loan for it...  so it was built over 10 years piecemeal
20:33.31Ariel_Assid, I only use them outbound for 800 there stil at 2.4 cents but there a backup
20:33.59Assidi use nufone for everything and vp as backup..
20:34.11Assidtoll free = toll free on NF
20:34.17tzangerthe banker asked him why he never got a loan from him in those 10 years...  my 2nd cousin said "Actually I have to thank you.  If I'd have played your game I'd have had a mortgage and owed on the house still... now instead I own my own house and have plenty of cash."
20:34.21Ariel_tzanger, I have land paid for in the Ocala area I want to build a house. But my money is all tied in the house I live in how. Sucks.
20:34.41tzangerAriel_: yup... not fun
20:34.58Ariel_nufone has had problems last week or so for me to canada. So I have had to add there areacodes to vpc
20:34.59AssidAriel_: sell house.. get money.. go build new house.. live there
20:35.08Assidaah
20:35.18Ariel_Assid, yes it's on the plan but I have to live someplace while i build
20:35.37Assid'girl next door'
20:35.43funxionlol
20:35.45*** join/#asterisk TechJournalist_ (n=chatzill@d141-133-19.home.cgocable.net)
20:36.04spackleOne of my friends unclues lived in a tent/trailer while he cleared his land and built his house
20:36.11spackleDon't think I'd reccommend that tact.
20:36.23Assidno more asterisk box
20:36.44extremisariel_: can a regular voicepulse account handle many simul inbound calls?
20:36.55Ariel_I have a wife, and 2 girls that live with me... (kids) so can't do that. But was thinking of a trailer home while I build
20:36.55epabloWhere should look for nat issues with a asterisk box on a certified IP and clients on NAT?
20:37.06Assidextremis: i think 4 simultanous
20:37.13Ariel_extremis, I have been able to get up to 4 calls
20:37.33Ariel_epablo, sip debug
20:38.08*** join/#asterisk TechJournalist (n=chatzill@d141-133-19.home.cgocable.net)
20:38.18epabloAriel_: What am I looking for?
20:38.22Assidokay.. for nce.. i will sleep early today...
20:38.29Assidmaybe 1/2 hr
20:38.56Ariel_epablo, to see were your rtp is sending the stream to ip address if not then ethereal to see what is happening
20:39.01extremisAriel_: what kind of account should I get? does the typical byod account support iax? It appears to require a MAC
20:39.15epabloAriel_: Ok.. thanks.
20:39.20Ariel_extremis, connection.voicepulse.com
20:39.41Ariel_it's a byod it's on the little menu at the bottom of there site as well.
20:40.03Ariel_I have to get going see you later folks.
20:40.12Uther_Pl8rz
20:40.17Assidlater Ariel_
20:41.54*** join/#asterisk klictel (n=klictel@207.107.208.140)
20:44.10*** join/#asterisk MiJaMu (n=mmullen@cdm-24-248-221-79.bnvl.cox-internet.com)
20:45.14MiJaMuhello, all
20:45.45MiJaMudoes anyone know where to go for more info on agi and learning the basics of it?
20:48.29*** join/#asterisk arguile (i=user224@66.38.201.234)
20:48.47redder86how loud can silence be?
20:49.19klictelso loud you can't hear it
20:49.21arguileredder86: about 40db on my ATA given the gain settings
20:49.42dos000anyone can share their experience with nicely priced did numbers ?
20:49.57redder86arguile: how does 40db translate into a 16-bit slinear value?
20:52.26azzie_dos000, how much are you willing to pay? :)
20:52.53tzangerexcellent
20:52.59tzangerI've secured two computers for asteirsk testing
20:52.59dos000azzie_, the lower the better ;-)
20:53.10tzangerI can attack this Zaptel DTMF issue again and the TDM400 interrupt hogging
20:53.13redder86because in analyzing received silence from the PSTN I'm seeing 16-bit slinear values ranging from 0x10 through 0x84, and I'm wondering how lenient I should be in my definition of "silence".
20:53.27tzangera P3/700 for the TDM and a Duronsomething for the TE405
20:53.33dos000azzie_, this is for north america btw ... europe would be icing on the cake
20:53.54azzie_dos: bunch of providers offer that...
20:53.55redder86I do see 0x8 for a minute
20:54.17azzie_dos: some of them even for free, if you don't care about area code ;-)
20:54.26dos000azzie_, i was trying to get a pointer to places that review those or even personal experiences
20:55.03azzie_dos000, look through forums at dslreports.com for personal experience ...
20:55.04redder86ah, and one instance of 20ms of 0x00 (true zero-energy)
20:55.07dos000azzie_, which ones are free ? i mean i need something that can be called from a normal fone btw
20:55.25*** join/#asterisk logicalonline (n=Ken@209.242.52.25)
20:55.45dos000azzie_, read pstn .. i am wondering how they can offer free !
20:55.54klicteldos000 we have PRIs and pay between $5 and $10 a month per did, not including any minutes of course
20:55.56azzie_dos000, ipkall.com
20:56.10azzie_dos000, gives you a did in Washington state
20:56.47logicalonlineanyone here using the directed pickup app successfully?
20:57.01dos000klictel, is there free incoming on did anywhere ?
20:58.04dos000klictel, or even flat rate
20:59.08klicteli don't beleive there is anything free anywhere
20:59.18klicteli dunno about dids
21:00.32epabloI don't think you will find free DIDS
21:00.48niZonhas anyone ever used chan_alsa?
21:00.57azzie_epablo, ipkall.com
21:01.28dos000epablo, not free .. just flat rate included in the monthly bill. i am hoping someone already figured this
21:02.01niZonstanaphone also has free DIDs
21:02.05niZonincoming
21:04.10epabloWhere I work we have flatrate on the incomming but You have to pay for the DID
21:04.18dos000niZon, ok lemme rephrase it. I need a wholesale did numbers with flat rate pricing.
21:04.29dos000if that even exists !
21:04.49niZonoh
21:05.11epablodos000:  We have that with 954, 305 and 561 numbers and Sao Paulo numbers
21:05.12azzie_dos000, how many minutes per month?
21:05.17clive-klictel do you use iax2 trunking?
21:05.52dos000azzie_, incoming ?
21:06.16azzie_dos000, we have any did's but with paid incoming calls
21:06.54dos000azzie_, no wholesale or flatrate pricing on incoming ?
21:07.24azzie_dos000, nobody likes to give flat-rate to businesses ;) they tend to abuse the service too much
21:08.13dos000dos000, this is going to be resold to costomers .. so teknically speaking one can set treshhold for hoe and busniss usage.
21:09.54swm_~bang dos000
21:10.03swm_~beat azzie_
21:10.04jbotACTION beats azzie_ with a large stick.
21:10.15Beirdo~trout azzie!
21:10.19swm_~beat his dick
21:10.23jbotACTION beats his dick with a large stick.
21:10.32Beirdo~seen pussy
21:10.41jboti haven't seen 'pussy', Beirdo
21:10.45Beirdoheheh
21:11.00dos000azzie_, care to discuss private ?
21:11.19azzie_dos000, nothing with flat-rate comes up in mind, sorry :)
21:11.30swm_~beat his giant dick
21:11.31jbotACTION beats his giant dick with a large stick.
21:11.45*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
21:12.43azzie_swm: too much coffee?:)
21:14.31*** part/#asterisk logicalonline (n=Ken@209.242.52.25)
21:15.01*** join/#asterisk |Vulutre| (n=Vulutre@111.220.204.68.cfl.res.rr.com)
21:16.00dos000azzie_, whats the company/website btw ?
21:16.23|Vulutre|can anyone recommend a good IAX/SIP provider for LD, VPC is being very crappy
21:17.02denonno complaints with nufone
21:17.16denonand they serve up iax and/or sip
21:17.32|Vulutre|does nufone do automatic refills yet?
21:17.41syleif you don;t want to pay nufone rates , i found voipjet really good
21:17.57denonof your account? nope, though I think you could schedule a paypal to recur
21:18.34sylei don;t think so denon, i think you have to setup IPN code on your website to handle billing people
21:18.35|Vulutre|hmm okay
21:18.50niZonsomeone correct me here, chan_oss is supposed to output to the sound card
21:18.58denonno clue -- I just stick in as much as I need, and add more when I need more :)
21:19.06sylei haven;t done paypal php coding in awhile so maybe its changed
21:19.34Hmmhesays<PROTECTED>
21:20.10*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:20.45*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
21:22.20eKo1anybody used Sayson phones?
21:22.57*** join/#asterisk redax (n=redax@catv-50621cc0.catv.broadband.hu)
21:23.03redaxhi,
21:23.21*** join/#asterisk santiago (n=santiago@63.245.86.145)
21:26.17*** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net)
21:26.31ronaldl79G'Day, folks.
21:27.22ronaldl79How many of you guys are using * exclusively at home for call termination? (No Vonage, CallVantage, etc.)
21:27.51pauldyI am
21:27.52arguileI am
21:28.11ronaldl79Is everyone pleased with the performance?
21:28.35pauldyso far so good I'm not a heavy phone user
21:28.43ronaldl79I have deployed a few PBXs, and also use it at home -- but have had Vonage for over two years for my 'day-to-day' dialing.
21:29.24arguileronaldl79: I actually have four DIDs coming in due to the number of people in the house, very happy with IAX to DID provider even on DSL
21:29.47ronaldl79Now, I want to use Asterisk full-time -- since I think it just makes since. So, any recommendations on termination and DID providers? I have used VoicePulse, and have VoipJet for termination -- but perhaps there's something better? I also need a provider who instantly provisions DIDs, because I need one for Denver, Detroit and Nashville.
21:29.58ronaldl79Nice, arguile.
21:30.18pauldyronaldl79, I use broadvoice for all call services
21:30.19ronaldl79sense*
21:30.40ronaldl79How are they, pauldy? I've read some things. Just wish I could find an unlimited provider -- incoming/outgoing.
21:30.43pauldywhen I added a did to my account via their web interface I was able to dial it imediatly and get to my pbx
21:31.06ronaldl79But how much are you paying per DID, and which plan do you have? And, can you place multiple calls at once?
21:31.57tzangerronaldl79: really look at the figures... unlimited never is and with a per-min provider with the right rates you can end up saving significant money anyway
21:32.14pauldyI think it was 1.95 I cut the original one, plus 22.95 for unlimited I'll look it up,  and I have had six simultanious calls running at once using meetme
21:32.22*** join/#asterisk RickTick (n=rpulido_@c-67-191-89-108.hsd1.fl.comcast.net)
21:32.52arguileAnd if you go per minute look at the initial billing and billing interval
21:33.10pauldyI have unlimited world
21:33.17arguileMost aren't true per second. I'm on 30/6
21:33.52pauldyI have used well over 6000 minutes to california in my second month with no issue
21:34.58pauldysorry my bill is 22.09 total
21:36.50ronaldl79Sorry guys...trying to get Vonage to unlock my Cisco ATA.....
21:37.50redder86anyone here got a fax machine that I can send a test fax to?
21:37.51ronaldl79paudly ... so you're termination multiple calls via Broadvoice, eh?
21:37.52niZonhas anyone worked with chan_oss?
21:38.02ronaldl79terminating, rather
21:38.26ronaldl79paudly -- Are you pleased with Broadvoice? How's the voice quality?
21:38.32pauldyyup I did this a while back for a month
21:38.35ronaldl79pauldy -- Any setup fees?
21:38.51pauldydoing a ton of conference calls and such
21:38.54*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
21:38.58ronaldl79Cool
21:39.04ronaldl79So a $1.95 per DID, per month?
21:39.06pauldyit is ok I didn't really have much to compair it to
21:39.19pauldyits a lot better than a cell phone
21:39.38pauldybut I wouldn't says it is the same a t-1 and possibly slightly worse than pstn
21:39.42ronaldl79Anyone else care to chime in on Broadvoice or something better?
21:40.45sylei heard people had issues with broadvoice
21:40.49sylein this channel
21:40.50pauldythe setup was 10 dollars
21:41.01ronaldl79Yeah, I have too, syle....just browsing online...
21:41.17ronaldl79Vonage surely better unlock my ATA!!!!!!
21:41.29syleof course they won;t
21:41.38ronaldl79I'm still waiting on the phone.......
21:41.39sylei heard people were able to J-tag it open though
21:41.43pauldysyle, I think that becuase there are so many people moving to broadvoice that is %1 has issues and comes in ehre looking for help it just seems like a lot of peoples having problems
21:41.53pauldyI haven't had a problem yet
21:42.29ronaldl79Vonage is trying to charge me a termination fee of $41 for the ATA ... saying they'll refund it when I return the ATA ... BS ... I've been a subscriber since 2003 ... I'm waiting to hear word on that fee and unlocking my ATA
21:43.01ronaldl79Correction, they haven't *tried*...they already did...lol
21:44.14pauldyI though vontage always owned their terminal adapters
21:45.07pauldyoh BTW ronaldl79 there is a section for ISP reports at broadbandreports.com and it has a listing with ratings for companies that provide VOIP
21:46.06*** join/#asterisk Tangent (n=Arc_Tang@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk)
21:46.14ronaldl79Okay, I got the termination fee removed by Vonage....
21:46.30ronaldl79However, they're telling me that my ATA isn't locked .... and that I can do anything with it....BS
21:46.54ronaldl79I know this thing is locked....I can't even see it via the web or telnet.
21:47.21arguileI'd really like one of the Linksys WRTP54Gs they use, the -NA unlocked model of course
21:47.49ronaldl79Freedom of choice is best ... and being locked to a service is BS....
21:47.50pauldythere is a nice jtag connector on those if you have the skills to solder on a pin header
21:47.52azzie_ronaldl79, have you read their terms & conditions ?
21:48.06ronaldl79Not recently, azzie....
21:48.06pauldyronaldl79, you have freedmom of choice
21:48.08arguileWith modified firmware, better QoS, and asterisk all on the appliance... :)
21:48.17pauldyone choice you use existing hardware
21:48.26pauldythe other you use new hardware
21:48.35ronaldl79At least with having access to this ATA, I don't have to run out and buy a SIP phone just yet....
21:48.38azzie_ronaldl79, they always had must-return-ATA policy
21:48.52azzie_ronaldl79, same as broadvoice and [most all] other providers
21:49.16ronaldl79azzie -- I recall after a certain subscription period, that it no longer applied. Thus, that's why they removed the termination fee for the ATA.
21:49.23pauldyI think the sip phones are a good investment
21:49.25*** part/#asterisk Uther_P (n=uther_p@66.180.120.82)
21:49.32pauldyesp with the features they come with now
21:49.35ronaldl79They are, Pauldy, I just don't have the cash to get a nice phone righ tnow.
21:49.46ronaldl79I'm looking at one of the Uniden IP phones.
21:50.05ronaldl79the UIP200 and UIP???? wireless
21:50.11pauldyI got a cheap gxp-2000 and I'm pretty happy with it
21:50.13arguileI have one SIP phone, but most of the time the $19.99 analog models win out :)
21:50.28ronaldl79How's that, arguile?
21:50.43azzie_ronaldl79, like over a year or more...
21:50.55*** part/#asterisk epablo (n=epablo@200.75.139.188)
21:50.56arguileronaldl79: The phones? Price :)
21:50.58pauldyyou mean they decide to die on you
21:51.00ronaldl79Right, azzie, and I've been a subscriber since February 2003....:)
21:51.34arguileAnd existing residential wiring as oposed to rewiring for cat-5
21:51.39niZondoes anyone use chan_oss?
21:51.46niZonI can't seem to get any sound
21:51.47ronaldl79I don't, niZon, sorry.
21:51.49niZonmpg123 works fine
21:52.47ronaldl79Did anyone see the pics of Von Fall 2005? Digium has a big booth.....
21:53.39ronaldl79pauldy -- Does BroadVoice provide CID w/name incoming to your PBX?
21:53.42*** join/#asterisk sozo (n=vidar@pppoecl74009.minlos.no)
21:54.07pauldyso far most all calls ahve had appropriate cid info
21:54.16ronaldl79with name?
21:54.21pauldyyes
21:54.31pauldyI'm sure there have been a handfull without
21:54.35ronaldl79And it reflects on your soft and hard phones, and CDR?
21:54.44distortionname isnt handled by service provider
21:54.50pauldybut whne they aren't provided it gives you the location of the co
21:54.54*** join/#asterisk TechJournalist (n=chatzill@d141-133-19.home.cgocable.net)
21:55.17pauldyso I might get a call from GRANDPARIE,TX 214502xxxx
21:55.24ronaldl79cool
21:56.27pauldyI use it when the gate outside my townhome calls me to let people in
21:56.37pauldythen run a script to just let them in
21:56.46ronaldl79Cool.....
21:57.15*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
21:57.17ronaldl79I think * is the best thing since pizza....so many things u can do with it....I'd like try out that bluetooth presence thingy, too...
21:57.30pauldythought about adding a layer of auth to it but I've lived here for 4 years and no one has called me via the gate I wasn't expecting to call me from the gate
21:58.26niZonronaldl79: or chan_oss
21:58.45ronaldl79chan_oss is related to sound, right?
21:58.54pauldyhehe nizon what are you doing with the chan_oss trying to make a PA
21:59.05niZonyeah
21:59.14niZonbut i don't get any audio
21:59.18pauldyhave you checked voip-info.org
21:59.18sozoI'd like if bristuff would work with CVS HEAD...
21:59.23niZonit just says it opened the channel blah blah
21:59.24niZonyeah
21:59.35niZonvery few results on google
21:59.54pauldyno luck, I've never done it myself don't plan to anytime soon
22:00.00niZonhm
22:00.27pauldyfrom what I saw though you can just set it up in your extentions.conf and roll on
22:01.55pauldythere is a way to describe chan_oss from the command line
22:02.03pauldyin asterisk
22:03.17niZonyeah i know
22:03.21niZoni got it partially working
22:03.31niZonwith the Dial(CONSOLE/dsp stuff
22:03.33pauldyniZon, what distro are you running?
22:03.38niZonand it answeres and everything
22:03.45niZoni'm using that xorcom package
22:04.01pauldywhat kernel is it?
22:04.05pauldyuname -a
22:04.34niZon2.6.8-2-686
22:04.41pauldyif what I'm seeing is right you may want to load the alsa module instead of oss
22:04.48chiardonxorcom . . . .from delta 345! galaxy
22:05.10niZonthing is, if I load alsa it doesn't seem to work at all
22:05.17niZoni just get unknown channel type
22:05.22pauldyhave you tried setting nooad => chan_oss.so
22:05.23ronaldl79Why is Vonage tech support clueless about unlocking a damn Cisco ATA 186 device????
22:05.29niZonyes
22:05.30pauldyand then load => chan_alsa.so
22:05.37niZonnot the specific load
22:07.12FuriousGeorgeso how come the conference button on my eyebeam is greyed out?
22:07.14ronaldl79Is there not a factory reset for the Cisco ATA 186?
22:07.27pauldycause you don't have two people on the line
22:07.39pauldyboth having answered
22:07.58FuriousGeorgepauldy: duh!
22:08.00FuriousGeorge:)
22:08.32niZonwith alsa: app_dial.c:777 dial_exec: Unable to create channel of type 'CONSOLE'
22:08.48pauldywow neat
22:10.19ronaldl79Now Vonage wants to charge $15 just to unlock my OWN ATA!!!!!
22:10.28ronaldl79This is comical!
22:11.56FuriousGeorgeits my understanding that QoS is out of my hands once my data leaves my network.  what i was wondering is:  cant it prioritize outgoing traffic so i dont saturate the upstream on my calls?
22:12.34pauldynow you send them a bill as a contractor for 200 with a description of brokering repair of cusotmer ATA
22:13.02pauldyFuriousGeorge, for the most part yes
22:13.11*** join/#asterisk BBHoss (i=HydraIRC@68-191-105-222.dhcp.dctr.al.charter.com)
22:13.18BBHosshello all
22:13.34niZongah I'm hungry and i have my ccna class soon
22:13.39BBHossi need some help
22:13.57BBHossvoipex is giving me the runaround and won't get me my info
22:14.14BBHossanybody got a contact #?
22:14.20pauldyniZon, can you run the alsamixer
22:14.32*** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net)
22:14.51niZonnope
22:15.09*** join/#asterisk darkskiez (n=darkskie@86.132.171.33)
22:16.15pauldychange it back to oss then
22:16.16pauldyhehehe
22:16.34*** join/#asterisk Abbas (i=Abbas@203.81.223.239)
22:17.08niZonyeah i did
22:17.46*** join/#asterisk Abbas (i=Abbas@203.81.223.239)
22:18.10niZoni shall bbl
22:19.06*** join/#asterisk leku (i=ekim@ip68-14-27-152.ri.ri.cox.net)
22:19.06lekuhola
22:19.06lekuare there any whitepapers out there comparing asterisk to cisco callmanager?
22:19.12JerJeryeah, "Don't use CCM"
22:19.28JerJerthat's black text on white background
22:19.28lekukeke
22:19.31lekufor real though
22:19.57lekuI have a very limited understanding of the two products and I want to know what the differences are
22:20.07lekuI am not even sure if they are in the same league
22:20.27JerJerthen don't ask me, i am very biased
22:20.55chiardonbiased to what? JerJer
22:21.22lekuwhat is "pdpc" ?
22:21.23JerJerpick a topic
22:22.18JerJerPremier's Drug Prevention Council
22:22.28JerJerProcess Decision Program Chart
22:22.36JerJerPeer-Directed Projects Center
22:22.40JerJerpick one
22:22.41*** join/#asterisk doolph (n=doolph@200.46.148.34)
22:22.44eKo1~pdpc
22:22.52eKo1nada
22:22.54lekun=JerJer@pdpc/supporter/bronze/jerjer
22:23.05lekuok whatever
22:23.07JerJerthat's freenode
22:23.15JerJeror the orginization behind freenode, rather
22:23.30lekuanyone know about PSTN simulation?
22:23.51JerJerlike simulated traffic?
22:23.52JerJerload testing
22:24.04lekuI want to setup a complete end to end voip lab
22:24.15JerJerhope you got $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$
22:24.17JerJernot just $
22:24.18lekuyes
22:24.33lekuyour tax money hard at work bra
22:24.58*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
22:24.59JerJerthen hook me up with some - I am about to go to Hawaii to test asterisk   :P
22:25.08lekuwhere in hawaii?
22:25.24JerJersome university lab, i believe
22:25.26lekuoh
22:25.36lekuthought you were working for the navy or something
22:25.40JerJerbut its for a magazine
22:25.57JerJerwell not officially at least
22:26.18eKo1bastard
22:26.44*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
22:27.03JerJerAvaya and Cisco will have their crap there too
22:27.06JerJerso it will be fun
22:28.10redder86I've already tested Asterisk in Hawaii :-P
22:28.31redder86it works fine there when the power is up :-D
22:28.58pauldyniZon, you still there
22:29.11pauldyI just setup mine for using oss nad it produces no audio as well
22:29.28pauldyI'm wondering is moh doesn't have something to do with it
22:29.58FuriousGeorgei noticed terrible sound when i transfered a call from a cell to another cell with all voip in between.  my theory is that it has something to do with a combination of low gain and heavy echo cancelation.  does that sound accurate?
22:30.21*** part/#asterisk leku (i=ekim@ip68-14-27-152.ri.ri.cox.net)
22:30.25*** part/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
22:30.35FuriousGeorgemy understanding is that mobile providers use heavy echo cancelation
22:31.00opus_like a women screaming? :)
22:31.56FuriousGeorgewell, sure thats one source of echo
22:32.04redder86or pain
22:32.17eKo1both
22:33.08FuriousGeorgeat first i thought bandwidth was the issue but i figured that since im using ulaw and 2 conversations on a 376kb dsl connection
22:33.14FuriousGeorgethat cant be it
22:33.28eKo1latency maybe
22:34.28FuriousGeorgeeKo1: i would expect with latency just a long delay b/w statement and response on the other end.  this sounded broken up, and uninteligible
22:34.50FuriousGeorgebut what do i know
22:35.07darkskiezI want to file a bug report but I dont want to encourage the wrath of 'not a bug' but it is a serious bug, but I cant reproduce it, but other people have had the problem. Whats the best way to handle this.
22:35.21darkskiezor wrath of 'technical support issue'
22:35.25*** join/#asterisk perlmonky (n=perlmonk@69-168-21-26.chvlva.adelphia.net)
22:35.25RoyKfscking bugs
22:35.37RoyK#3986 drives me mad
22:35.39*** part/#asterisk perlmonky (n=perlmonk@69-168-21-26.chvlva.adelphia.net)
22:36.13eKo1darkskiez: fix it
22:36.17darkskiezis there not a patch ?
22:36.24eKo1make one
22:36.24darkskieztheres like 20 patch files for that one
22:36.44darkskiezeKo: i'm looking at the code now, but i dont know asterisk internals well enough..
22:36.59darkskiezplus I cant reproduce it to test the code
22:37.48eKo1hehe, i'm facing similar issues with a module of mine
22:37.58darkskiezI added a bit of text explaining it to the bottom of a wiki page someone made: http://www.voip-info.org/tiki-index.php?page=Ring+requested+on+channel
22:37.59eKo1and i know asterisk internals
22:38.15syleare gas prices going to go down?
22:38.39sylejust had some dude show up at my house saying he could lock me in at my current rate for 5 years
22:38.58sylekinda like an adjustable vs fixed mortgage kinda thing
22:39.47darkskiezI put an offer in on a flat today
22:39.53darkskiezdont talk about mortgages
22:40.13JerJerwho wants to buy a VoIP provider?    cheap
22:40.26darkskiezyou selling one? is it in debt?
22:40.27sylehow many customers?
22:41.14RoyK~seen oej
22:41.21jbotoej <n=Olle@64.251.117.114> was last seen on IRC in channel #asterisk, 9h 48m 26s ago, saying: 'Cool'.
22:41.31RoyKthis is NOT cool
22:41.37RoyKARGH
22:41.39JerJersyle:  last i counted over 10,000
22:41.59azzie_jerjer: active and paying ones?
22:42.01RoyKJerJer: selling your company?
22:42.09sylePM
22:42.38opus_Furious are you using IAX or SIP?
22:42.41opus_~seen drooth
22:42.43jbotdrooth <n=drooth@118.56.77.83.cust.bluewin.ch> was last seen on IRC in channel #asterisk, 8d 14h 16m 2s ago, saying: 'i want to tell you something in our channel'.
22:42.50opus_8 days whoah
22:42.56RoyK~seen Jesus
22:42.57jbotjesus <i=jesus@84-122-34-139.onocable.ono.com> was last seen on IRC in channel #debian, 28d 4h 28m 26s ago, saying: 'hi all'.
22:43.05RoyKlol
22:43.06opus_whoah
22:43.15JerJerRoyK:  i would want double the best offer if you bought it
22:43.18azzie_jerjer: i heard vonage ads cost around $200 to get one customer, so maybe that's the way to go :)
22:43.22RoyK~seen Elvis
22:43.23jbotelvis <~elvis@9-151.tr.cgocable.ca> was last seen on IRC in channel #kde, 150d 5h 16m 3s ago, saying: 'StevenR, thx a lot bro'.
22:43.31opus_azzie really
22:43.32JerJer150 days lol
22:43.41opus_azzie where did you read that?
22:43.54azzie_opus: i know it's insane
22:44.01RoyK~seen Mohammed
22:44.04jboti haven't seen 'mohammed', RoyK
22:44.10RoyKbummer
22:44.13*** join/#asterisk modulus_ (i=modulus@rm-f.net)
22:44.15modulus_bleh
22:44.15RoyK~seen Mohammad
22:44.17jbotmohammad <~samy@62.3.46.58> was last seen on IRC in channel #tacobeam, 449d 8h 8m 10s ago, saying: 'Keefak?'.
22:44.17JerJer~seen allah
22:44.18jbotJerJer: i haven't seen 'allah'
22:44.22modulus_sup roy
22:44.24modulus_sup jerjer
22:44.31JerJernobody else has seen allah either
22:44.35azzie_opus: but if you watch TV in some metro area and see how MUCH ad they run...
22:44.45modulus_thanks
22:44.47modulus_*muncH*
22:44.47modulus_*muncH*
22:44.48darkskiezeKo1: you dont happen to know, say, given a pri channel, how to force close the channel that thinks it is open?
22:44.56JerJermodulus_: SSDD
22:44.58RoyKJerJer: that's just arabic for God, same guy...
22:45.04opus_azzie so if you multiple their customer base by $200, thats there marketing budget?
22:45.13RoyKas with Jhv
22:45.19JerJerum no that god is not my god
22:45.21RoyKor Jahve or Jehova or whatever
22:45.24modulus_jerjer, FTYVM
22:45.39RoyKsame books
22:45.39azzie_opus: i'm not good at sales calculations and i don't like to speculate; i just told what i heard from my sales ;)
22:45.41RoyKsame stories
22:45.42RoyKsame gods
22:45.44RoyKor god
22:45.51*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
22:47.11RoyKanyone here wanna get $1k for fixing a bug?
22:47.16modulus_sure
22:47.28modulus_which bug?
22:47.30RoyK#3986
22:47.35eKo1darkskiez: you can always kill it
22:47.38*** part/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
22:47.39JerJerwho's definition of a bug do we use?
22:47.41eKo1err, destroy it
22:47.48*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
22:47.50RoyKops
22:47.51RoyKhttp://voip-info.org/tiki-index.php?page=Bug+3986+Bounty@
22:47.56modulus_royk, are you sure it's not a feature?
22:49.10RoyKmodulus_: honestly, i doubt that
22:49.13opus_Whoah, $1k for that bug?
22:49.17opus_shit
22:49.21RoyKyes
22:49.24RoyKfor me it's major
22:49.30opus_its because there ain't no protected code block grabbing ports.
22:49.31modulus_it's a major bug fo ryou?
22:49.38RoyKwe've got one patch from an .it guy that didn't work
22:49.49opus_modulus - the problem is that asterisk only uses 1% of the available million ports
22:49.58modulus_that's an easy fix
22:49.59opus_RoyK whats your setup
22:50.16modulus_tcp/ip ports?
22:50.16JunK-YRoyK: for u, everything is major, which is false.
22:50.16RoyKSIP-IAX2 gateway with a few thousand SIP clients
22:50.23RoyKJunK-Y: please
22:50.31RoyKJunK-Y: this is really critical
22:50.40modulus_udp ports?
22:50.46RoyKJunK-Y: not the average annoyence
22:50.47eKo1maybe you should make it $5K
22:50.49RoyKmodulus_: yes
22:50.55modulus_iirc there are only 65535 ports, not a million
22:51.32darkskiezget the ipv6 patch :)
22:51.44RoyKmodulus_: yes. and if it climbs above 10k in less than 24 hours, there's no point increasing to 50k, is it?
22:51.54redder86https://sourceforge.net/projects/iaxmodem/
22:52.05darkskiezmore ports? use a second ip
22:52.07pauldy8*65535 for tcp
22:52.21modulus_pauldy, yes yes, and you know what i meant
22:52.28pauldyerr. for ip
22:52.31pauldydoh
22:52.36RoyKthe point is. there's a leak there......
22:52.43RoyKand i want that fixed
22:52.49modulus_what's the leak?
22:52.57RoyKRTP ports leak
22:53.06modulus_it's not worth $1k for me
22:53.07RoyKso anyone wanna make $1k in a hurry, go on
22:53.10modulus_it's worth $2k
22:53.24darkskiezpauldy: whered u pick the 8* from ?
22:53.32RoyKit's probably done with a few hours of hacking
22:53.33FuriousGeorgehas anyone found a decent usb headset device intended for telephony (as opposed to videogames)  i used to use these earbud bluetooth things, but they're meant for cell's and a hassle
22:53.37*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241)
22:53.44RoyKPoWeRKiLL: morning
22:53.49modulus_royk, tell me more about the environment
22:54.02FuriousGeorgeplus when you teach the users to use them, it turns out they connect to pure static from time to time
22:54.03spackleFuriousGeorge, Does Plantronics make one yet?
22:54.20PoWeRKiLLEvening RoyK
22:54.21RoyKsimply SIP->IAX. the leak seems to increase on incoming calls from SIP, which are handled by AGI
22:54.32RoyKPoWeRKiLL: 1am here, so, well, ok
22:54.36FuriousGeorgespackle: the blue tooth earpieces i tried were plantronics and they had the same issues as the generic ones i tried before them
22:54.38pauldydarkskiez, there is a sinlge byte field in ip that defines UDP/TCP/Etc.
22:54.55pauldyso you can have ports that range across all of those types
22:55.08darkskiezpauldy: aye,  u said 8*65535 for TCP tho
22:55.18modulus_royk, is it a matter of asterisk re-allocating used ports that are disconnected?
22:55.32pauldydarkskiez, read there is more than just that one line
22:55.34spackleFuriousGeorge, So no plantronics USB sets yet?  Too bad.  Logitech makes some
22:55.58FuriousGeorgespackle: i got some of those, but they look like "air traffic control"
22:56.02RoyKmodulus_: well. the ports stay open when they should be closed. it doesn't even help restarting asterisk with a 'restart now'. i need to shutdown asterisk and then restart it
22:56.13PoWeRKiLLRoyk :)
22:56.14modulus_that's some horrible code
22:56.15darkskiezpauly: not all IP protocols have ports, do that doesnt make sense.
22:56.21darkskiezs/do/so/
22:56.24RoyKpauldy: TCP and UDP run on top of IP
22:56.27spackleFuriousGeorge, Good looks and functionality, why don't you just throw the moon in there too.
22:56.28FuriousGeorgespackle: i looked for usb one eared gooseneck type headsets for ever
22:56.29RoyKIP doesn't have ports
22:56.31RoyKUDP and TCP does
22:56.50FuriousGeorgespackle: its not just looks, if you dont have a free ear its hard to hear people not on the phone
22:56.52darkskiezICMP doesnt, GRE doesnt .. etc
22:56.56modulus_man you guys make me feel smart
22:56.59modulus_S-M-R-T
22:57.00modulus_SMART!
22:57.02RoyKICMP has ports IIRC
22:57.10modulus_it's like an ego boost every time i come in here
22:57.17darkskiezreally, should really read the rfc again
22:57.19modulus_royk, if i'm given $2k i'll fix it
22:57.20darkskiez(i should)
22:57.22*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
22:57.29PoWeRKiLLI still didn't find my zap solution on my TE410P
22:57.51modulus_powerkill, i have had no problems with my te410p
22:57.54SimonRdoes anyone have G729 compiled on their machine?
22:57.54RoyKmodulus_: $1k should do
22:57.54modulus_i have isdn pri
22:58.01RoyKmodulus_: it's not up to me
22:58.03modulus_royk, i'm worth more
22:58.05FuriousGeorgethe best i have found, but havent actually tried, are these one eared headset devices made for the play station
22:58.06SimonRMy network is down because after reinstalling the machine, the codec_g729.so file is gone
22:58.12RoyKmodulus_: also, i have a guy working on it
22:58.19pauldyanyway there are 8 siffernt protocols that have been defined to have ports
22:58.24darkskiezroyk: doesnt look like it does.
22:58.31pauldywhich is why I said 8 not 256 you argumenative bastards
22:58.33SimonRPlease please please, if someone can find it on their machine and DCC it to me, I would be eternally grateful. Just type locate codec_729
22:58.34pauldy:-P
22:58.53modulus_ok simonr, i typed: locate codec_729
22:58.54darkskiezmore to the point you can have more than one IP on the interface
22:59.06RoyKdarkskiez: hm. no. i was wronng
22:59.20darkskiezso you can have squillions*8*65536 :)
22:59.32modulus_# locate codec_729
22:59.32modulus_ksh: locate:  not found.
22:59.32modulus_#
22:59.33RoyKanyone else want to earn $1k for a few hours debugging?
22:59.40modulus_simonr, done.
22:59.44*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
22:59.48darkskiezi'd love to, but as long as its not RTP bugs.
22:59.48*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
22:59.48RoyKsince modulus_ wants to push the price higher and higher.......
22:59.56modulus_my time is very valuable
22:59.59RoyKdarkskiez: hehe
22:59.59darkskiezand its bed time, night.
23:00.03SimonRmodulus, could you please email it to simon@directleap.com?
23:00.08modulus_i just posted it
23:00.10modulus_here:
23:00.12modulus_# locate codec_729
23:00.12modulus_ksh: locate:  not found.
23:00.12modulus_#
23:00.16*** join/#asterisk websae (i=websae@207-118-153-15.dyn.centurytel.net)
23:00.17RoyKbrb
23:00.19modulus_i apparently don't have locate
23:00.29modulus_of course i am trying it on an RS6000 AIX machine
23:00.31SimonRfind / -name *729
23:00.35SimonRfind / -name *729*
23:00.37Cresl1nisn't it in /var/lib/asterisk/modules?
23:00.38SimonRthat will also work.
23:00.40SimonRves
23:00.46websaeanyone know of a good VOIP company that I can become a wholesale partner with?
23:00.47Cresl1nor /usr/lib/asterisk/modules
23:00.51Cresl1nI can't remember which
23:00.56SimonRthe filepath is /var/lib/asterisk/modules/codec_g729.so
23:00.57*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
23:01.15SimonRI have 6 boxes down until I can get a copy of this file...
23:01.29modulus_damn that sucks
23:01.32PoWeRKiLLAnyone already got this problem SIP -> * -> ZAP -> PSTN the audio is passed to the SIP Device even when ringing so all call are mark as answered in cdr
23:01.37PoWeRKiLLany idea ?
23:01.56websaehrm
23:01.59websaeno ideas here
23:01.59modulus_yeah use my customer app_cdr i wrote
23:02.02modulus_custom*
23:02.03websaedon't have that setup
23:02.28websaehow about anyone know of VOIP providers that give you service at wholesale to sell to others?
23:02.55rayvdwe use PacWest in northern california for that websae
23:03.14websaewhat do they provide to you?
23:04.16distortionwebsae, what are you looking for?
23:04.20spacklefailure to plan on your part, yada yada yada
23:04.43websaeI am looking for a VOIP company....i want to sell VOIP to customers...i want to get it at wholesale cost
23:04.58RoyKmodulus_: apt-get install slocate
23:04.59RoyKperhaps
23:05.27distortionemail me: chris@comsolo.com perhaps we can help you
23:05.35pauldyjerjer is sellinghis company
23:05.45Corydon-wwebsae: why wouldn't a voip company bypass you and just go to the customer directly?
23:05.49spackleno way
23:05.56RoyKpauldy: don't pay more than $1k
23:05.57RoyK:P
23:06.02pauldyhahaha
23:06.04Corydon-wIf you can't answer that, you're not going to be in business for long
23:06.11redder86JerJer is selling?
23:06.22websaewhy bypass me when I can bring in customers...
23:06.30pauldyI'm not interested in voip for anything right now but saving money on ld
23:06.51rayvdhttp://www.pacwest.com/products/pstn/
23:06.52pauldyand doing geeky stuff with my asterisk box
23:06.54rayvdthere's what we're using websae
23:07.02websaedistortion: what kind of deal?
23:07.03distortionweb: this nick isnt registered, sec
23:07.04rayvddoesn't look like wisconsin is covered ;)
23:07.08Corydon-wwebsae: and why would those customers go to you, when they can go direct?
23:07.20Corydon-w(and save money)
23:07.25SimonRDoes anyone have codec_g729.so compiled based on the instructions on the web here? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/doc-050903.txt
23:07.26websaebecause I am their IT consultant company
23:07.38pauldyso you want to make a little residual
23:08.32websaecorrect
23:09.04websaewell i keep have clients ask me about VOIP services...residential and businesses...and I am trying to figure out how I could make money with that...
23:09.49sozoSimonR: yeah. got it working here but then I have tried to get either bristuff or mISDN or something working with my * CVS HEAD and now g729 doesn't work anymore
23:10.14SimonRsozo: do you have the old binary file?
23:10.26opus_simonR I have an installer that compiles it.
23:10.27redder86websae: you make money by getting a T1 and colocate an Asterisk server
23:10.35sozoyeah
23:10.37SimonRopus_ how does that work?
23:10.38*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
23:10.51UberbotHi all.
23:10.52opus_SimonR : you run a script, and tada..
23:10.59SimonRWhere can I find the installer?
23:11.06SimonRCould you email it to me?
23:11.11krisguyhello everyone
23:11.52opus_I am selling it for $25.00
23:11.53sozoi followed the steps and got it working pretty easily I'd say
23:12.21sylewhat is the best way to pass back variables to asterisk from agi script
23:13.04opus_or one pizza
23:14.43redder86If you're into bleeding and want to use HylaFAX with true PSTN lines connected to Asterisk... https://sourceforge.net/projects/iaxmodem/
23:14.54krisguymmmm, pizza
23:16.35opus_HylaFAX the PCI card?
23:17.41redder86opus_: huh?
23:18.27*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
23:19.40*** join/#asterisk shyru (n=shyru@dsl-201-128-19-18.prod-infinitum.com.mx)
23:19.42generalhancan anyone help me out with a couple of error messages? ive never seen these in my console before today. and they have popped up about 10 times since this morning
23:19.45generalhanWARNING[2399]: chan_sip.c:4826 check_auth: Stale nonce received from 'Break Room <sip:break@192.168.0.46>
23:20.01generalhanNOTICE[2399]: chan_iax2.c:6198 iax2_poke_noanswer: Peer 'voicepulse-wgw001' is now UNREACHABLE!
23:20.28generalhanwhat the hell is Stale Nonce ? lol
23:21.08RoyKmodulus_: still not interested in looking into that bug? it might be just a few lines, and for $1k, that should be worth it IMHO
23:25.49pauldyI'm interested in a module that detects and deals with fax over sip
23:26.45rayvdi wish you weren't :(
23:26.47*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
23:26.51redder86pauldy: I think that was added to NVFaxDetect some time ago.  That said, unless the path between the fax source and your Asterisk server is completely controlled you probably shouldn't bother.
23:27.15RoyKpauldy: that's called t.38 and there's a bounty for it
23:27.19modulus_bleh
23:27.32RoyKmodulus_: ?
23:27.41*** join/#asterisk HolyGod (n=free@206-248-146-24.dsl.teksavvy.com)
23:27.42distortionmmm t.38
23:27.49redder86T.38 is different than fax detection.
23:28.03redder86You could pass T.38 through SIP if you wanted to.
23:28.21modulus_royk: ?
23:28.29RoyKredder86: yep. but fax over SIP will need t.38 unless you're on a very low-latency network, as in LAN
23:28.46redder86RoyK: that is correct
23:29.31redder86Fax over IAX works pretty good over the localhost adapter. :-)
23:29.40redder86no latency there
23:30.28modulus_jbot jbot?
23:30.29jbotmethinks jbot is dumb
23:30.36modulus_jbot are you an infobot?
23:30.37jbotyes i am
23:30.42modulus_jbot weather klax?
23:30.59pr0mi faxed something today over iax2.  the fax machine said it sent "OK".
23:31.15spackleyou beginner luck
23:31.18pr0msent it over the internet through voicepulse connect.
23:31.19redder86pr0m: you had a fax machine connected to an IAXy?
23:31.39redder86pr0m: how many hops between you and voicepulse?
23:31.52pr0mthe fax is connected to the house phone wiring which is then connected to a digium card.
23:32.02pr0mhmmm.  dunno.  lemme check.
23:32.46RoyKjbot weather oslo
23:32.59RoyKjbot: lart himself
23:34.04modulus_jbot insult RoyK
23:34.44pr0mredder86: 16 hops total from my asterisk box which is behind the outside router on my lan.
23:35.02redder86pr0m: and did you confirm that the receiver got it?
23:35.26pr0mummmm. no.  actually. i made that assumption, cause i didn't get the call back.
23:35.33*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
23:35.40redder86hrmm... well, then I'd say that you were lucky.
23:35.50pr0mhowever, i can confirm that i received a fax from office max the other day...
23:35.55redder86I've accidentally tried faxing several times through NuFone with no luck ever.
23:36.10redder86well, that's a lie, it did actually complete a time or two
23:36.14pr0mi used NVBackgroundDetect app to convert it to tiff and send it to me by email.
23:36.31pr0mit's legible.
23:36.39redder86and that office max fax came to you through voicepulse?
23:36.54pr0mi'll test a fax with you if you want.
23:37.13pr0mredder86: yes.  over the internet from voicepulse connect.
23:37.50pr0mi just set this up last week so i can't be sure of much except what i've received so far from my test at office max.
23:38.20*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
23:38.31xhelioxNOTICE[2199]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 3! --- I keep getting that, what is it and should I be concerned?
23:43.19*** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer)
23:47.07pauldyok anyone know if chan_oss is broke or something in 1.0.9? I can't seem to get it to load up properly
23:47.21JunK-Ywhich error?
23:47.33pauldyI have headphones setup and I can stop asterisk and use mpg123 to listen to the mp3 file sin the moh dir fine
23:47.55pauldyI setup and extension and the oss.conf file and tried calling it but I'm hearing nothing come out the Cnsole/DSPp
23:48.00pauldyerr dsp
23:48.34pauldyno error just OSS/dsp answered SIP/200-7435
23:48.57pauldybut no audio then I hangup and everything looks normal
23:49.09pauldyrunnign asterisk -c -vvvvv -dddddd
23:54.25shyruhi!!
23:54.29shyruguys!!
23:55.42*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
23:56.32redder86JerJer: someone said you were selling?
23:57.38shyrui have 2 spa 2000 and 2 spa  841, the spa 841 does ok dial in and dial out , but the sip 2000 imposible dial out does not run
23:57.40shyru:S
23:58.04JerJerto the lowest bidder

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