irclog2html for #asterisk on 20060421

00:00.39orlokHmm, these netcomm ata's are shitting me
00:00.40Zodiacalanyone know why my cisco 7960 hardphone can see the status (hint) of my cisco ip communicator softphone and vise versa. but my hardphone can't see the status of another hardphone? i have sccp v7.2.4 firmware and chan_sccp feb 04 version...
00:01.13litagein sip.conf, does it matter if the order of allowed codecs is different between the [general] context and a [trunk] context?
00:02.39*** join/#asterisk apardo (n=apardo@87.217.146.193)
00:03.44justinu|laptopsorta
00:05.09justinu|laptopit'll use the order you specific in [general] when it can't find any matching peers in your other sections
00:05.14justinu|laptops/specific/specify
00:06.37distortionits supposed to use the order as the "preference" in the sip peer as first priority and then in the general section if not specified in the sip endpoint, although i havent seen it work very well.
00:07.09distortionsome reason ive only had luck with locking the codec to one specific.
00:07.38justinu|laptopyou'll have trouble if the UA you're talking to call indicates it supports a codec that you don't have
00:07.52justinu|laptopbut in general you're right
00:07.54justinu|laptopit sucks
00:08.15*** join/#asterisk MacDome (n=eseidel@A17-255-100-59.apple.com)
00:08.20distortionyeah, the trouble with boarder controllers (nextone/sansay/ect) is that they include a lot of codecs in the sdp.
00:08.38*** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl)
00:09.09distortioni should say the trouble with asterisk and SBCs rather.
00:09.12justinu|laptopwhy would they do that?
00:09.29Z_Godanyone knowledgeable about chan_modem here?
00:11.41justinu|laptopdone with work for the day!
00:11.42justinu|laptopw00t
00:11.49distortionnice.
00:12.02b4kaquestion, when i start asterisk with zapata.conf configured with my channels it fails to load chan_zap complaining of not finding device for chan ##
00:12.12b4kaif i start with an empty zapata.conf it works
00:12.30b4kaand i notice that in /dev/zap/ there arent any channel device
00:12.34b4kalike in my other asterisk
00:12.47*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
00:12.58b4kathis one im talking has the wan card, but itsnot plugged
00:13.03b4kais it normal then?
00:13.21justinub4ka: what distro
00:13.39chiardon!!!ubuntu!!!
00:13.51b4kafreebsd
00:13.54justinuoh
00:13.55b4kawith zaptel-bsd
00:14.03Nuggetyay bsd.
00:14.06justinuyou need to create the /dev/zap devices manually, perhaps
00:14.16b4kai have some devs there
00:14.17justinunugget probably knows tho :)
00:14.22b4kathe imging and zap* shit
00:14.34b4kabut in my linux box there are other devs with numbers
00:14.37Nuggetmy freebsd asterisk server is zaptelless.
00:14.38b4kathe channel numbers
00:14.50justinueven ztdummy-less?
00:14.54b4kawich asterisk complain arent there
00:14.55Nuggetyes
00:15.12b4kamaybe if i start the zapata device with the wan plugged in
00:15.16b4kait will create them?
00:15.34justinushould work without anything being plugged in
00:15.42justinudid you setup /etc/zaptel.conf correctly?
00:15.46b4kathat's what i think
00:15.47b4kayes
00:15.52b4kaasterisk starts
00:15.56b4kawithout zapata.conf
00:16.03b4kaan empty file
00:16.07b4kawhen i put my configured file
00:16.09justinuztcfg -vv works?
00:16.09b4kawith channels
00:16.11b4kayes
00:16.20justinunot sure... i never worked on bsd before
00:16.30b4kaits a sangoma card
00:16.38justinuyou might ask them
00:16.41justinuthey know what's up
00:16.41b4kaso i have wanpipe connecting to zapata
00:17.13*** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net)
00:17.24b4kawhat devs you have in /dev/zap?
00:17.41justinua bunch of channels
00:17.47b4kaoh
00:17.49justinupseudo, timer, ctl, channel
00:17.50b4kanumbers?
00:17.52b4kayeah
00:17.52justinuyeah
00:17.58b4kaim missing those...
00:17.59b4kadamn it
00:18.02justinuthis system is wanpipe also
00:18.05justinubut linux
00:18.08b4kathey are on the linux one, that is working now
00:18.24b4kaim going to upgrade the machine and put a freebsd
00:18.46justinutry creating the device nodes manually, perhaps
00:18.46b4kayou have an AFT card?
00:18.53justinuno, it's a A101
00:19.03b4kaand how do i know what nods it uses?
00:19.09b4kaah
00:19.11justinuwell, the major node is 196
00:19.15b4kai have an A102
00:19.21justinuand the minor nodes is the channel ID
00:19.26b4kahow do you know that?
00:19.46justinuit shoes the major, minor numbers when you do ls -l
00:19.59b4kayes
00:20.04b4kabut dunno if its the same on bsd
00:20.10justinuis the major node for your channel device 196?
00:20.18b4kaim not at work now
00:20.22justinuheh
00:20.29justinussh?
00:20.30b4kai just got home
00:20.34b4kabut it still bothers me
00:20.42b4kaermm, we have a pretty tight config
00:20.46b4kano nothing from outside
00:20.51justinuno remote access is a real PITA
00:21.04justinui think i'd have to quit that job
00:21.15b4kaheh
00:21.34b4kasecurity is a concern
00:21.41b4kawhen you're a is company
00:21.58justinuwhat happens if something goes down and you're not there?
00:22.04b4kayou go :)
00:22.10justinuwhat if you're out of town?
00:22.17b4kayou're not out of town
00:22.21justinufuck that
00:22.24b4kaand if you really are... someone else goes
00:28.02*** join/#asterisk novu5 (n=novu5@adsl-221-127-178.rmo.bellsouth.net)
00:31.48*** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
00:31.51*** part/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
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00:32.46*** part/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
00:33.10*** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
00:45.46*** join/#asterisk captsulu32 (n=captsulu@62.0.118.70.cfl.res.rr.com)
00:46.05captsulu32hello
00:47.14captsulu32Have question I need to know if asterisk can be setup to record all the calls in wmv/mp3/wav files
00:47.25captsulu32any ideas
00:51.51wunderkinmp3 or wav, yes, show application monitor
00:53.26*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
00:53.51*** join/#asterisk SplasPood (n=jwb@ool-18b93e04.dyn.optonline.net)
00:58.31captsulu32that's from the CLI interface right?
01:00.15[hC]so is there a win32 call manager / caller id popup interface of preference?
01:00.17[hC]Im just about to try out ADM
01:03.29*** join/#asterisk mogorman (n=4692d428@68.62.165.168)
01:04.40*** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
01:05.01*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
01:07.41*** part/#asterisk novu5 (n=novu5@adsl-221-127-178.rmo.bellsouth.net)
01:08.00*** join/#asterisk supjigatr (n=syslod@152.53.17.26)
01:08.31supjigatrShould faxdetect=both turn off echocan on a fax?
01:12.02Zodiacalanyone know why when i dial *97 with chan_sccp it asks for the mailbox number insted of automaticly knowing it?
01:12.25drfoomod2is anyone familiar w/ SS7?
01:17.17Kattyhi lads.
01:17.34drfoomod2evening
01:18.03Netgeeksevening Katty
01:18.59*** join/#asterisk Gershwin (n=fake@63.250.233.162)
01:19.59GershwinIs there a CRM, CMS or groupware type app that's generally considered to have very good asterisk integration?
01:20.33NetgeeksSugar CRM supposably has some asterisk integration modules
01:20.35Gershwinhmmm, scratch CMS... not what I was getting at
01:21.14GershwinYes, I've tried Sugar and was initially impressed.  However, my IT review board (inc. CEO) was not
01:21.18*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
01:21.24Netgeeksdoh
01:21.28supjigatrdfoomod2: Yes
01:21.29Gershwin...and upon hearing their criticisms, neither was I... :/
01:22.13*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:22.48GershwinI just did a search for "integration with asterisk" and got quite a few hits... this should take me a while
01:23.14Gershwiniotum... hmm.. reading...
01:23.40marcus2anyone here using chan_jingle yet?
01:26.53chiardonAnyone knows what's the use of chan_phone.so?
01:30.12dlynesHeya Katty...having a better day than you were this morning?
01:30.28*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
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01:36.10Drukenhmm..... 9:30 already....
01:36.16*** join/#asterisk jsaunders (n=root@216.86.121.58)
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01:37.38dlyneseasterner!
01:37.49Druken?
01:38.02Drukenuhmm... central ontario thank you...
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01:38.20Drukeni'm no fucken easterner....
01:38.25dlynesheh
01:38.29dlynesyou're east of thunder bay :)
01:38.35dlynesThat's the eastern half :)
01:39.02Drukennah.. that's the middle :)
01:39.18Drukeneastern starts after quebec :)
01:39.24dlyneslol
01:39.24Qwellafter quebec?
01:39.32drfoomod2EDT!
01:39.34dlynesIf you ask a quebecer
01:39.38dlynesNothing comes after them
01:39.44dlynesThey're the most important province
01:40.03alephcomWhy don't they want to be Easterners?  They sure don't want to be Westerners
01:40.05Drukenyou got that all backwards.... nothing comes BEFORE them... hehe
01:40.22xachenNow I know why I hate this province
01:40.32Gershwinoh, you're not from PEI?
01:40.39xachenno
01:40.42xachenI'm from ALberta
01:40.47xachenand backed up even longer for highspeed
01:40.52GershwinI thought only two things came from Alberta
01:40.53dlynesWhat's wrong with alberta?
01:40.55alephcomis xachen slamming AB??
01:40.56xachenwhen I needed it like 3 weeks ago
01:41.12*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
01:41.12GershwinWell, at least you've got Stampede coming up eh?
01:41.19alephcomxachen: and another tower's going live here next week.  So close, and yet so far
01:41.25dlynesxachen: that's not alberta's problem...it's more because Telus is a pia
01:41.25xachenheh
01:41.27Drukennothin wrong with alberta
01:41.29xachenalephcom: Les is an idiot
01:41.33xachenhe had a dish on our roof
01:41.41xachenand a hill was "skipping" a signal
01:41.41drfoomod2is * all good w/ SuSE?
01:41.43Gershwintower... as in WISP tower or something else?
01:41.46xachenyeah
01:41.46Drukenhehehe i like telus
01:41.56Drukenhow do ya know i'm in ontario? hehe
01:42.08Gershwinhe prob. whois'd you
01:42.09xachenand now he can put me on his other tower that is perfect
01:42.12xachenbut trees are in the way
01:42.13Gershwinand recognized your hostmask
01:42.13dlynesdrfoomod2: if it's good with Linux, it should be good with SuSE...if it's not good with SuSE, SuSE's got a problem
01:42.19xachenand he doesn't have time now til the 3rd of May
01:42.22xachenam I pissed?
01:42.24xachenbeyond
01:42.30xachenat this rate EVDO would even work ;P
01:42.39Gershwinwhat speed can you get from this provider?
01:42.46xachenabout 500 Kbits
01:42.47DrukenGershwin: no, that's one of those, ya know i'm not from the west because i actually like telus
01:42.56dlynesDruken: lol
01:43.11GershwinHmm, 500 kbits doesn't sound all that special unless it's the only thing going
01:43.14Drukeni've yet to find a evdo network...
01:43.16drfoomod2dlynes: pt. well taken
01:43.19dlynesDruken: Mind you....Telus is about 100x better than Group Telecom
01:43.29Drukenpisses me off, i paid a mint for the god damn aircard
01:43.35drfoomod2dlynes: i was looking at their Carrier Grade Linux compliant version
01:43.49dlynesDruken: Group Telecom/Bell is biggest pain in the ass to deal with...you have to go through twenty levels of bureaucracy to get anything done
01:43.55Gershwinwell, at work, we pay $350 US/mo for 6M/6M wireless
01:44.16Gershwinwhich is pretty donkey spankin good if you ask me
01:44.25Drukendlynes: well, gt is now rogers, and bell is like trying to get sex out of a 12 year old
01:44.36dlynesdrfoomod2: it's probably a 2.6 kernel with low latency optimizations or something
01:44.45alephcomGershwinL  We'd jump on that if we could.
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01:44.48dlynesDruken: GT is Bell, not Rogers
01:44.55dlynesDruken: Fido is Rogers
01:45.05Drukenmaybe there, but in ontario, gt went to rogers
01:45.11dlynesOh
01:45.28Drukenactually, gt went to sprint, and rogers purchased sprint
01:45.47dlynesI guess maybe Bell grabbed it here so they could get a stronger foothold in BC, and said forget the ontario base because they already have a strong presence there
01:46.02Drukenya think?
01:46.18dlynesWell, Bell was virtually non-existent here before they got GT
01:46.29dlynesBut it's still operating under the GT banner
01:46.34Drukensame reason why telus purchased bctel
01:46.43dlynesTelus and BCTel merged
01:46.49dlynesNobody bought anyone
01:47.05Drukenwell, reguardless, it's all the same now...
01:47.20dlynesYeah, and it's no worse than it already was
01:47.28dlynesTelus sucks just as bad as BC Tel did
01:47.33Drukeni'm still trying to get used to talking to chicks from my own city when i call customer service....
01:47.39*** join/#asterisk Dorphalsig (n=asterisk@200.71.58.39)
01:47.40Drukensince they opened up a call center here
01:48.05drfoomod2dlynes: it's way more than that.  checkout http://www.osdl.org/lab_activities/carrier_grade_linux
01:48.46drfoomod2dlynes: click on Registration on the left navbar
01:50.04drfoomod2dlynes: and if you realy want to kill yourself, these are the actual requirements for CGL 3.2: https://www.osdl.org/lab_activities/carrier_grade_linux/requirements.html/document_view
01:50.47dlynesdrfoomod2: cool
01:52.29drfoomod2np
01:52.52drfoomod2dlynes: i have a carrier grade server and I was looking for CG linux distro's
01:53.07drfoomod2the ultimate would be to take * to CG status
01:53.18*** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
01:54.24dlynesdrfoomod2: Yeah...not gonna happen without including SER or OpenSER In the mix though, I don't think
01:54.45dlynesdrfoomod2: Asterisk doesn't support load balancing and/or clustering afaik
01:54.58DorphalsigHi
01:55.13DorphalsigI have a TE400P card
01:55.14Gershwinso, considering Open/Free/NetBSD is out of the question drfoomod2?
01:55.22drfoomod2dlynes: what about http://www.voip-info.org/tiki-index.php?page=Asterisk%20High%20Availability%20Solutions
01:55.40drfoomod2Gershwin: not sure I understand
01:55.52Nuggetasterisk can be made to function reasonably well on freebsd, and passably well on any other posix-based system.  but I wouldn't suggest it for someone who is new to asterisk.
01:56.23Nuggetif for no other reason that you'll come here for help and people will ignore you because "it works in linux" even if the problem has nothing to do with your platform choice.
01:56.24GershwinI'll take your TE400P and raise you two Dialogic D480/SCs and 1 Adtran TA750
01:56.26drfoomod2i played w/ FreeBSD at some pt. in the past but the #freebsd turned me off
01:56.51Gershwinyes, I agree, operating systems should be judged by ones initial experience in an OS IRC channel
01:56.54Nuggetfor non-asterisk use, of course, freebsd is clearly the most superior solution for all needs and anyone choosing any other unix is clearly a moron.  :)
01:57.12drfoomod2his $210 ta750
01:57.25drfoomod2Gershwin: well then there ya go
01:57.55Katty...
01:58.11GershwinFor example, I now know that Python is a terrible language because someone called me a 'dooty head' in there a few months ago
01:58.17dlynesdrfoomod2: yeah...don't know...every time I hear somethign about someone wanting to do failover with asterisk everyone always suggests SER
01:58.22drfoomod2Gershwin
01:58.25Gershwinso the same applies to all applications, OSs, languages, etc...
01:58.29drfoomod2Gershwin: chuckle
01:58.36Nuggetheh
01:58.59Dorphalsigwhen I do modprobe zaptel I get the following error TE4/0/1 TE4/0/2 TE4/0/3 TE4/0/4 is dead with no revival
01:59.03drfoomod2Gershwin: it must have been a bad day
01:59.06DorphalsigWhy do I get that?
01:59.18DorphalsigAnyone?
01:59.21dlynesNugget: I didn't find FreeBSD to be as stable as Linux for working with Asterisk and the pri cards
01:59.31NuggetI don't either.
01:59.46*** join/#asterisk hads (n=hads@203.109.245.87)
01:59.48GershwinYa, use whichever you can get working reliably or have expertise with
01:59.48dlynesNugget: Not to mention the updates for the drivers and asterisk seem to not be terribly timely for freebsd
01:59.55NuggetI don't even run ztdummy on my freebsd asterisk box any more.
02:00.10NuggetI don't know what you mean about "and asterisk" there, though.  asterisk is asterisk.
02:00.14dlynesNugget: So are you just using it for a softswitch then?
02:00.17Nuggetthe release schedule is the same for everyone
02:00.37NuggetI don't need meetme or music on hold.  for everything else it seems to work fine enough.
02:01.06dlynesNugget: I was referring to /usr/ports/net/asterisk
02:01.11DorphalsigJustin?
02:01.14DorphalsigAnybody?
02:01.30Nuggetall packaged asterisk sucks.  the gentoo port, the freebsd port, any rpm you find.
02:01.36Nuggetjust compile it yourself.
02:02.02paolob_Hi guy! I can't get a pap2 connect to a asterisk server. Any hint? Any web page to read? thank you!
02:02.07dlynesNugget: I do, for slackware..I was told to use the ports tree for freebsd by the guy i work with that's a "freebsd expert"
02:02.13Dorphalsiga good way to have the benefits of * and rpm would be to make your own rpm
02:02.20Nuggetnot always.  some ports (like asterisk) aren't all that great.
02:02.23Dorphalsigthat way you can uninstall / upgrade easily
02:02.28alephcomThat's exactly what the freebsd port does.  It compiles it localy.
02:02.29Nuggetapache1 for example.  the apache1 port is abysmal.
02:02.43alephcomI guess if the port patches have problems then it's not so great
02:02.47Nuggetalephcom: no, it does all kinds of other needless crap and it's stale
02:02.51Dorphalsigjustinu|laptop, ... you there man?
02:03.17Nuggetasterisk used to require massive patching to build on anything but linux, and the freebsd port is full of vestigial cruft from those long-gone days.
02:03.20dlynesNugget: Not to mention it applies the bri patch, installs openh323 (why, I don't know...it shouldn't be part of asterisk), mpg123 (again, not necessary), ...
02:04.22alephcomIt's tough to keep those from getting stale...  I'm not jealous of the maintainer.
02:04.41dlynesI don't mind using FreeBSD for a generic Internet server, but I don't think i'm going to use it for an asterisk server again any time soon
02:04.53Nuggetthat would be my advice
02:05.21NuggetI hate linux as much as the next guy, but I put up with it for asterisk and gridmp.  sometimes you're just locked in.
02:05.31b4kaheh
02:05.40b4kai have 1 linux in my 30 servers
02:05.45dlynesBesides the fact that I don't know anything about making binary packages for it, and i'm not terribly familiar with the system discovery tools under freebsd is a huge hindrance to me for adopting it for asterisk use
02:05.45b4kaand im decided to kill it
02:05.53b4kathe only linux is the asterisk server
02:05.56tier_1so is postgress only supported by odbc ?
02:05.58b4kanext week, it'll be gone
02:06.23dlynestier_1: no, but the postgres module hasn't been updated in a while
02:06.40dlynestier_1: the odbc driver is actually better than the postgres driver for postgres
02:06.50b4kalinux should stay in the desktop market...
02:07.02dlynesLinux has a desktop market?
02:07.03b4kathats how i use it
02:07.11b4kayes it does
02:07.13tier_1so is postgress only supported by odbc ?
02:07.21tier_1sorry
02:07.24dlynestier_1: Did you not read what I just said?
02:07.34b4kawell
02:07.36b4kai do too
02:07.39tier_1I hit the wrong key
02:07.43dlynesah
02:07.56b4kabut osx86 its still strange
02:08.05b4kaand i dont have SSE3 in my machines
02:08.09b4kanor SSE2
02:08.11theorem_anyone here have a HP jetdirect j2591a or similar printer sharing device ?
02:08.29b4ka?
02:08.31DorphalsigI have a TE400P card amd I'm running zaptel 1.2.5 When I do modprobe zaptel I get the following error TE4/0/1 TE4/0/2 TE4/0/3 TE4/0/4 is dead with no revival
02:08.35Nuggetosx86 isn't legit.
02:08.51b4kaer, who cares?
02:08.53NuggetI do.
02:08.56theorem_Nugget - last I checked OSX ran on x86 machines
02:08.56b4ka:/
02:08.57dlynesDorphalsig: is that before or after you've loaded the te4xxp driver?
02:09.06SwKtheorem: its called piracy
02:09.13b4kamy only legal copy of software is my starcraft
02:09.13NuggetI think piracy is a pretty vile and shortsighted thing to engage in.
02:09.15Dorphalsigbefore ...
02:09.25Dandanlol
02:09.30Dandani am like 95% legal
02:09.30theorem_b4ka - you're an idiot for saying that
02:09.37b4kawhy?
02:09.41theorem_esp. in a software dev. channel.
02:09.44Dandanjust probably two pieces of software on my pda is not
02:09.53b4kain a FREE software channel?
02:09.56theorem_people make their living programming here.
02:09.57Dorphalsigdlynes, --> Before
02:10.06Nuggetyou're not talking about free software, you're talking about pirated softwae.
02:10.08theorem_not HERE ,but at work most likely
02:10.27b4kawell, theres a reality to confront
02:10.29Winkieanyone got a seen script? :)
02:10.31theorem_you're just not making friends.
02:10.32DandanFREE!=FOR FREE
02:10.33russellbjbot: Nugget++
02:10.34NuggetI believe very strongly that developer's have every right to choose how they license and distribute their work.
02:10.34*** part/#asterisk jason31337 (n=jason@c-67-169-212-156.hsd1.or.comcast.net)
02:10.37b4kasoftware is way to expensive here...
02:10.40dlynesDorphalsig: try modprobe wct4xxp instead
02:10.44DandanFREE AS IN BEER!=FREE AS IN SPEECH
02:10.51*** part/#asterisk captsulu32 (n=captsulu@62.0.118.70.cfl.res.rr.com)
02:10.51b4kano one buys legal software here
02:10.56dlynesDorphalsig: wct4xxp automatically loads the zaptel driver
02:10.58Nuggetno one?  I do.
02:11.00Dandanjbot: Nugget++
02:11.03theorem_.ar ?
02:11.05alephcomb4k4:  I do
02:11.06NuggetI always respect a developer's choice of license.
02:11.07theorem_where is that
02:11.08b4kaNugget, you live in argentina_
02:11.13b4kaand you too alephcom?
02:11.27NuggetI don't see how your country of origin would change the morality of the issue.
02:11.36Dandanindeed!
02:11.41b4kawhen you have to pay 5x what YOU pay
02:11.42theorem_b4ka - I suggest switching to a free as in beer OS ... your costs of software and legality will be cheaper
02:11.42b4kayou do
02:11.48alephcomb4k4: Canada
02:11.49De_Monwhere can I get meetme2?
02:11.52Dandanstealing is stealing, period
02:11.53Dorphalsigdlynes, --> I get an error... not beeing able to open /dev/zap/ctl
02:11.54Nuggetas you've pointed out, there are free alternatives if the price isn't justifiable
02:11.59b4kalol, piracy isnt ilegal here :P
02:12.09b4kaUS law doesnt play in the whole world
02:12.12theorem_piracy isn;t illegal ...
02:12.16Dandanisn't it? 3rd world country?
02:12.18theorem_b4ka - morals do
02:12.18Nuggetpiracy is not a US legal construct.
02:12.25b4kaindeed
02:12.32dlynesDorphalsig: Have you created that device node?  It should have gotten created when you ran make install on the zaptel drivers
02:12.39theorem_piracy is a slang term for stealing software
02:12.54b4kathere's no law for that here
02:13.01Nuggetpiracy is a 250 year old term to refer to the unauthorized reproduction of intellectual property.
02:13.13Nuggetthe word and concept have existed longer than the united stats has.
02:13.16Dandanlol, that is why all my servers have .ar in blacklist mode :)
02:13.27theorem_Dandan ?
02:13.27justinu|laptopi thought piracy came from sailing
02:13.37De_Monis it pirate day yet?
02:13.39Nuggetit also means that.
02:13.39b4kaDandan, i dont see how that helps you
02:13.41Dandantheorem_: firealls and email
02:13.55Dandanb4ka: .ar and .br have prolly the most bots I have seen
02:13.58Dorphalsigdlynes --> Just loaded zaptel and then wct4xxp. and I get the message again
02:13.59theorem_Dandan - ah I see.
02:14.01*** join/#asterisk rhett (n=heyrhett@cpe-72-177-11-82.austin.res.rr.com)
02:14.02Dandan(except for .ch ofc)
02:14.09*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
02:14.12rhett<PROTECTED>
02:14.12b4kaDandan, then you didnt see china?
02:14.18rhettcan someone give me an overview of how one would hypothetically set up asterisk to custom interactive voice response?
02:14.22Dandanand they are being used for SSH knocking and spam
02:14.24Dandan*.cn ofc
02:14.29b4ka:)
02:14.31Dandani didnt mean switzerland :)
02:14.44Nuggetswitzerland is .ch
02:14.48DandanI know :)
02:14.52Dandanit is a bit late
02:14.58Dandanand I had too many margies :)
02:14.58Nuggetoh, I see.
02:15.00dlynesDorphalsig: In your zaptel source directory, run the following command:
02:15.00b4kaanyway, the reality is that no one is going to pay 5000$ for a windows license
02:15.01Dorphalsigdlynes --> Strange thing ... I am getting "OK" on my E1s. But I'm getting yellow alarm on my channelbanks
02:15.02NuggetI missed the first line.
02:15.09b4kathat's more than a 2 month salary
02:15.09theorem_b4ka - is that the cost ?
02:15.11dlynesDorphalsig: make devices
02:15.17b4kayes it is
02:15.25theorem_b4ka - then don;t use windows
02:15.26theorem_:)
02:15.28dlynesDorphalsig: Does that run ok?
02:15.39Dandanwindows license has never been for $5000
02:15.43b4kaits not only windows
02:15.43Dorphalsigdlynes --> yeah install -m 444 bla bla bla
02:15.52theorem_they're merely shooting themselves in the foot for forcing costs that high
02:15.54b4kaDandan, 5000$AR
02:15.59dlynesDorphalsig: now type modprobe wct4xxp
02:15.59Dandanoh
02:16.02Dandanright
02:16.04rhettI want to set up an IVR app
02:16.16b4ka3:1
02:16.19rhettis there a good VOIP provider or something?
02:16.42theorem_b4ka - what is the symbol / name for your currency ?
02:16.45theorem_just curious
02:16.45*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
02:16.47*** join/#asterisk cced (n=dev2003@222.33.36.205)
02:16.49b4ka$AR
02:16.56b4kaARP$
02:16.58Dandanhttp://www.ezprezzo.com/crazypics/lego_aircraft_carrier.html OFF TOPIC :)
02:17.00b4kawhatever
02:17.11dlynesYeah...Argentina's banks went for a major shit a while back
02:17.12b4kaPESO
02:17.16Dandanrhett: all of them are good :)
02:17.17b4kayeah
02:17.19b4ka2001
02:17.24Dandandlynes: yeah, remember that
02:17.35theorem_Peso ?
02:17.39rhettHow do I set up a server that can call 100 people simultaneously?
02:17.44b4kayup
02:17.44Dorphalsigdlynes --> May I priv you?
02:17.52Dandanrhett: you gotta have 100 lines
02:17.53rhettand interact with their response?  How much does that cost?
02:17.54dlynesMost of the foreign banks pulled out of Argentina when that happened and the Argentina currency collapsed
02:17.58b4kaanyhow, i do support opensource software with donations...
02:17.59Dandanthat's 5 PRIs :D
02:18.02docelmoI want one!  Lego's RULE!
02:18.02b4kaim not that evil
02:18.09dlynesDorphalsig: sure
02:18.09Dandandocelmo: I liked it too :D
02:18.10theorem_"Top traders see peso strengthening to 48 to $1"
02:18.13theorem_that blows ..
02:18.23theorem_windows costs lets say .. $200 ..
02:18.24Dandanpeso used to be 1:1 to US$
02:18.29b4kayeah
02:18.30docelmoDandan I have 25 lego kits..  I collect them
02:18.37Dandantheorem_: WXP PRO 160 afair
02:18.40theorem_that was april 19th this year
02:18.42Dandandocelm0: I used to...
02:18.43docelmoI have them all over my living room.. My 6 year old likes to break them..
02:18.50theorem_oh, 16 0..
02:19.01rhetthow much does 100 phone lines cost?  There isn't a cheaper way to do it with voip?
02:19.01De_Mondocelmo what theme?
02:19.02Dandanlol coool
02:19.08theorem_$7680  PESOs argentinian
02:19.10rhettIs that a lot of bandwidth?
02:19.11theorem_that's rediculous
02:19.14Dandani gotta get myself one of those '6 years old' :D
02:19.22Dandani al just trying for now
02:19.23b4kaheh
02:19.23theorem_how much is say ... a gallon of milk ?
02:19.31Dandanrhett: planning on using voip?
02:19.33b4kawe dont use gallons
02:19.35b4kaso i dont know
02:19.37theorem_or some other common food item ?
02:19.38Dandantheorem_: they use liters
02:19.44theorem_oh, right ..
02:19.47rhettI want to see what the cheapest way is, dandan
02:19.57Dandantheorem_: in economy there is a common denominator: "one bigmac"
02:19.57De_Monus is the ONLY country that uses that stupid system
02:20.02docelmotheorem_ get a phat internet pipe and voip the hell outa hit w/ asterisk
02:20.04b4kaanyway
02:20.10b4kafood is pretty cheap
02:20.13theorem_Dandan - unfortunately you are correct
02:20.21Dandanbig mac index: http://www.economist.com/markets/bigmac/displayStory.cfm?story_id=2708584
02:20.25b4kabecause the food is local
02:20.25theorem_outa hit ?
02:20.28b4kawindows isnt
02:20.30Dandanno, that is like economy 300 in college :)
02:20.54b4kabig mac is 3U#S
02:20.56b4kaU$S
02:21.08b4kathe meat is local, so the price doesnt go high
02:21.17theorem_right
02:21.22theorem_so, cheaper overall ..
02:21.28Dandanhttp://www.woophy.com/map/index.php
02:21.30theorem_but the conversion rate is killing you
02:21.30b4kawell
02:21.31Dandananother OFF TOPIC
02:21.36b4kacheaper if you have U$S :P
02:21.41theorem_yeah
02:21.55theorem_my $1  buys 46 or so Big Macs
02:21.59theorem_from AR
02:22.06b4kaermm
02:22.07b4kanot
02:22.16theorem_48 : 1 conversion rate
02:22.19DorphalsigI have a TE400P card amd I'm running zaptel 1.2.5 When I do modprobe zaptel I get the following error TE4/0/1 TE4/0/2 TE4/0/3 TE4/0/4 is dead with no revival. Could this be Zaptel's Watchdog fault?
02:22.27b4katheorem_: wth are you seeing?
02:22.34b4kait's 3:1
02:22.35theorem_http://money.inq7.net/topstories/view_topstories.php?yyyy=2006&mon=04&dd=20&file=1
02:22.45Dandanok i am off to bed
02:22.46Dandan:)
02:22.48Dandannite all
02:22.52Dandannite argentinian :)
02:22.58Dandan[d]
02:23.14theorem_"Veloso, treasurer at HSBC Philippines, has predicted that the peso will surge further to 47.50 to the dollar."
02:23.21b4kadude
02:23.26b4kathats from philipines
02:23.28b4kawtf?
02:23.31theorem_doh ..
02:24.09theorem_click on values to see graphs   1 ARS    in ARS
02:24.09theorem_<PROTECTED>
02:24.34theorem_ok, so 3 : 1 you're right
02:25.18*** join/#asterisk sloom (i=davew@cpc2-bagu5-0-0-cust974.bagu.cable.ntl.com)
02:25.58theorem_it looks like it recently took a turn for the worst ... jeeze .. it was nearly 1:! back in Jan !
02:26.05b4ka:S
02:26.14theorem_what happened ?
02:26.17b4kawell
02:26.23b4kait isnt just fro the worst
02:26.30b4kaexports are much better now
02:26.35b4kabecause we export in dollars
02:26.46b4kait was unrealistic to have 1:1
02:26.53theorem_true , everything is dirt cheap for everyone
02:26.53b4kawe were consuming money we hadnt
02:26.54*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
02:27.03b4kauntil it exploded
02:27.09theorem_exploded ?
02:27.12b4kayeah
02:27.16b4kathere was no more money
02:27.21b4kaand there had to be a change
02:27.23Nuggetinstead of pirating you should be running argentinian software (that isn't burdened with a tragic exchange rate) or open source/free alternatives.
02:27.55b4kai normally do
02:27.57theorem_I have t oside with Nugget on this one
02:28.02b4kai only have 1 windows
02:28.04theorem_it'll expose you to a different world
02:29.04b4kabut for example, getting a new game... would take 6months of wait... and it would be REALLY expensive
02:29.19b4kaand there's not other choice
02:29.31theorem_I see
02:29.33b4kabelieve me
02:29.35theorem_yeah ..
02:29.36b4kai would LOVE to buy it
02:29.37theorem_$30
02:29.45theorem_that'd be $90 ..
02:29.47b4kaand have the box
02:29.52b4kaand everything nice
02:29.53b4kaand shiny
02:30.03theorem_or more realistically $60 usd would be $180 ar$
02:30.12b4kanot reall
02:30.12b4kay
02:30.17theorem_yeah
02:30.18Nuggetdid you steal your computers too?
02:30.19b4kayou have to add shipping
02:30.25b4kaand other costs
02:30.25theorem_do you have a fast inet connection at leas t ? :)
02:30.47b4kaso 30$USD ends up $200ARS
02:30.55theorem_blech
02:31.00b4kai have a pretty decent con :P
02:31.27theorem_I'm willing to bet that it actually beats out the average speed here in the US
02:31.33b4ka20U$S 2mbits
02:31.36Nuggethow could you have possibly afforded to buy a computer in this bleak, insurmountable financial wasteland you're describing?
02:32.07theorem_Nugget - it was only recently that the USD to AR ratio went 3:!
02:32.07b4kasaving money
02:32.07theorem_it was previously nearly 1:1
02:32.09b4kait was 1:!
02:32.10b4ka1
02:32.11*** join/#asterisk pengyong (n=lala@221.6.132.220)
02:32.16Nuggetso saving money works fine for computers, but not for software?
02:32.33Nuggetmaybe you should save the kind of money that can buy both, and not just the dell dollars.
02:32.44b4kaNugget
02:32.50b4kahardware isnt that expensive
02:32.59b4kabecause they know they couldnt sell it otherwise
02:33.08mmlj4and good software is free :-)
02:33.19b4kai mean
02:33.23b4kai could buy 2 decent computers
02:33.27b4kawith 1 windows license
02:33.55b4kaas unrealistic as it sounds
02:34.21b4kaanyway, as i said, i use mostly freesoftware
02:34.25b4kaand i donate to them
02:34.27b4kawhat i can
02:34.39theorem_b4ka - ok you're not as bad as you originally came off
02:34.49theorem_finanaces do play a large role here
02:34.52b4kalol
02:34.54theorem_larger than I had imagined
02:36.05*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net)
02:36.22*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
02:36.26DorphalsigI have a TE400P card amd I'm running zaptel 1.2.5 When I do modprobe zaptel I get the following error TE4/0/1 TE4/0/2 TE4/0/3 TE4/0/4 is dead with no revival. Could this be Zaptel's Watchdog fault?
02:38.04paolobGuys, I'm trying to configure pap2 and asterisk with http://voxilla.com/index.php?name=News&file=article&sid=39 , but it doesn't work? Any hint?
02:38.09DorphalsigI have a TE400P card amd I'm running zaptel 1.2.5 When I do modprobe zaptel I get the following error TE4/0/1 TE4/0/2 TE4/0/3 TE4/0/4 is dead with no revival. Could this be Zaptel's Watchdog fault?
02:38.23justinu|laptopnever heard of that... sounds like a bad card maybe?
02:40.25*** join/#asterisk ledoktre (n=jon@209.152.67.6)
02:40.28ledoktrehello all
02:41.02ledoktretwo hopefull simple questions for anyone who's game ;-)
02:41.46DoktorGreggame
02:42.27ledoktre:-)  tanks.
02:42.32DoktorGregkk lets hear your simpleish question:)
02:42.38Dorphalsigjustinu|laptop, --> can ya give me a hand PLEASE?
02:43.01ledoktre1) I have 3 Sipura SPA-3000s.  How does a body create a "group" for outbound calls?  so it rolls to 2, if 1 is busy, etc
02:43.12tainted-Dorphalsig sounds like u need to buy a sangoma card next time
02:43.53paolobGuys, is the pap2 perfectly equivalent to the spa2000?
02:44.03Dorphalsigtainted-,  --> I'll bear that in mind. But for this issue ...
02:44.11*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-150-58-174.msy.bellsouth.net)
02:44.45*** part/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-150-58-174.msy.bellsouth.net)
02:45.57DoktorGreghttp://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
02:46.02theorem_paolob - well ... I was convinced it was by a seller on ebay ... supposedly linksys has stopped producing the sipura versions and are selling only the linksys ones at a slightly lower price
02:46.08theorem_oops
02:46.10*** join/#asterisk zeppelin_ (n=zeppelin@201.15.175.147)
02:46.11DoktorGregledoktre, look at section titled
02:46.18DoktorGregDialing a Group
02:46.46paolobtheorem_, I'm following the sipura2000 instructions, but I can't get the pap2 connected to asterisk... any idea?
02:47.01*** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
02:47.06nservsucksshow application digittimeout
02:47.18nservsucksnow that it's using set.. how do get get the information from the CLI?
02:47.26DoktorGregpaolob, whats your network topology look like?
02:47.43paolobDoktorGreg, a lan, with a asterisk server and a pap2
02:47.46DoktorGregnservsucks, asterisk -vvvvvr
02:48.14*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com)
02:48.28*** join/#asterisk MaxusSE (n=MaxusSE@62.213.40.155)
02:48.28DoktorGregstart with ping
02:48.35DoktorGregcan you ping the phone from the server?
02:48.42nservsuckshrm maybe I asked wrong
02:48.43nservsucksSet(TIMEOUT(digit)=timeout)
02:49.02nservsucksI'm using this now.. how to I get information about digit timeout on the CLI?
02:49.16paolobDoktorGreg, yes, I'm in the web interface configuring it
02:49.21DoktorGregkk
02:49.36DoktorGregnservsucks, to be honest i have no idea what you are trying to accomplish
02:49.38nservsucksI used to use "show application digittimeout", but now that it's in Set, how do I get information on its syntax, description etc?
02:50.00DoktorGregtype help at the asterisk cli
02:50.06DoktorGreggives you a list of commands
02:50.15DoktorGregand google
02:50.27MaxusSEHi!
02:50.43nservsuckstyping "help" at the CLI will show me how to use digit timeout in Set?
02:50.45DoktorGregthe www.voip-info.org wiki has the down low on asterisk programming
02:50.57nservsucksI'm aksing how to get that info on the CLI
02:51.07nservsucksi.e. show application type format
02:51.34nservsucksobviously you're not the correct person to ask
02:52.03nservsucksas the answer is either 1) it's no longer possible on the CLI or 2) it's "help $foo"
02:52.18file[laptop]show function timeout
02:52.20MaxusSEOn what OS for the best to installing Asterisk?
02:52.23ledoktreDoktorGreg, I just looked at the link.  I am not using zap, I am using 3 SPA3000's (sip) for outbound PSTN.
02:52.27NuggetMaxusSE: Linux
02:52.53file[laptop]er show function TIMEOUT
02:52.55nservsucksfile: No function by that name registered.
02:52.55file[laptop]it's case sensitive
02:53.00MaxusSEOn what Linuxs?
02:53.04NuggetLinux.
02:53.10nservsucksperfect!
02:53.14nservsucksthanks file
02:53.31file[laptop]show functions
02:53.49NuggetA 2.6 kernel can make some things easier, but it's not too important either way.  Just Linux.
02:53.51MaxusSENugget> SUSE - REDHAT - ........
02:53.55NuggetLinux.
02:54.50*** join/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net)
02:55.03MaxusSENugget> as on count FreeBSD?
02:55.09NuggetThat's not Linux.
02:55.34DoktorGregledoktre, best i can find suggests making a macro that dials each phone until it times out
02:55.40MaxusSEI know!
02:55.40DorphalsigAnybody can help me out configure my channelbanks?
02:55.59nservsucksAnyone know if there are plans to get an X100P card working on sparc64 architecture? :)
02:56.16MaxusSEbut even so
02:56.18nservsucksNot high on digium's priority scale I would guess :)
02:56.27*** part/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net)
02:56.33ledoktreDoktorGreg: thanks, Ill look into macros (shudders ;-) ).  question 2 - callerid. some users are blank.  is there ain if statement, if its blank, set=PRIVATE or something?
02:59.20MaxusSENugget> I have SUSE 10 , will go?
03:00.07NuggetThat is Linux.
03:00.34mmlj4* even runs on windows... I don't think the flavor of linux matters a whole lot
03:00.47DoktorGregMaxusSE, I think your need to reconsider all your life choices that you would choose suse over debian
03:01.05NuggetAny Linux is just as good as any other Linux.  Asterisk does not care in the slightest.
03:01.19NuggetUse whichever one you want, because it won't matter one iota for asterisk.
03:01.37b4kaor dont use linux :)
03:01.43NuggetI wouldn't suggest that.
03:01.49Nuggetespecially if you're not familiar with asterisk.
03:01.52xachen* works on BSD
03:01.52DoktorGregDebian is the one true linux!
03:01.53xachenjust not well
03:02.00b4kai even tried to get it working on openbsd
03:02.03b4kasilly me
03:02.09b4ka* only works in freebsd
03:02.15NuggetI run asterisk in linux, freebsd, and os x.  I wouldn't recommend it on anything but linux.
03:02.23MaxusSE<DoktorGreg> why
03:02.24b4kazaptel-bsd doesnt compile on open or net
03:02.36mmlj4um, it runs on netbsd, and openbsd, and bsdi (OSX), and ...
03:02.49DoktorGregnext thing you know people will start using vi instead of the one true editor, emacs
03:02.57Nuggetbarely, and not without plenty of challenges that you don't face in linux.
03:02.58mmlj4um, there are zaptel packages for netbsd, IIRC
03:03.04b4kanope
03:03.11b4kazaptel-bsd only compiles on free
03:03.15b4kathe author said so
03:03.32Nuggetyou can say "um" all you want, it doesn't change the fact that using asterisk on a platform other than linux can be a real hemorrhoid if you don't know what you're getting into.
03:04.10[hC]And aside from that, do you really care that much to run it on something else? Sadistic bastards. :P
03:04.13DoktorGregIm gonna agree with nugget
03:04.43DoktorGreg[hC], at some point OS choice becomes religion
03:04.51mmlj4b4ka: not to start a flamewar, but http://www.voip-info.org/tiki-index.php?page=Asterisk%20NetBSD
03:05.16DoktorGregbut the bsd'
03:05.30DoktorGregare not gnu compliant
03:05.39mmlj4why am I even defending this?
03:05.48b4kaoh, there's a zaptel-netbsd
03:05.51b4kadidnt know that
03:05.56b4kaanother fork YAY!
03:05.57mmlj4run what you bloody want to run
03:06.11dlynesmmlj4: because you've got too much free time on your hands?
03:06.25DoktorGregwill asterisk run under cygwin?
03:06.26mmlj4only until monday
03:06.36mmlj4DoktorGreg: yes
03:06.43DoktorGregomg thats sick
03:06.59dlynesDoktorGreg: not everything works for asterisk under cygwin though
03:07.07DoktorGreglike zaptel
03:07.08mmlj4hardware support is out
03:07.09DoktorGreg?
03:07.42dlynesDoktorGreg: I'm sure there's other stuff too...after all Windows isn't Linux
03:07.49DoktorGregI may dump windows and switch to osx for desktop
03:07.58DoktorGregbecuase i look around at my desktio
03:08.02DoktorGregdesktop
03:08.06DoktorGregand i use xchat for chat
03:08.11DoktorGregfirebird
03:08.14DoktorGregthunderbird
03:08.27mmlj4if all you're doing is VoIP (SIP, IAX2, etc.), it'll run moderately fine under cygwin
03:08.27DoktorGregseveral free OS databases
03:08.29dlynesFirebird is a desktop app?
03:08.51DoktorGreger firefix
03:08.57DoktorGregbut i use firebird as well
03:08.59mmlj4dlynes: part of the mozilla suite, i think it's the mail client
03:09.19DoktorGregfirebird is the old interbase OS project
03:09.21dlynesmmlj4: That's Thunderbird; Firefox is the browser, and Firebird is a database from Borland
03:09.46mmlj4hmm... ok
03:09.50DoktorGregbut ive been moving from delphi to lazarus as lazarus matures
03:09.58mmlj4lazarus?
03:10.03mmlj4never heard of it
03:10.08mmlj4is it free?
03:10.14DoktorGregyah
03:10.16DoktorGregand open
03:10.25mmlj4to me, that's the only criteria that matters, everything else is secondary
03:10.26dlynesmmlj4: And for what it's worth, Sunbird is the calendaring project
03:10.34mmlj4dlynes: i've used that
03:10.42mmlj4got webdav working with it, too
03:11.03DoktorGregbut all my webstuff is going a LAMP direction....
03:11.11DoktorGregall the little applications i maintain
03:11.18dlynesDoktorGreg: Lazarus is a pascal development environment for Linux?
03:11.21DoktorGregare going to lazarus
03:11.24DoktorGregyup
03:11.59DoktorGregpascal rocks!
03:12.17DoktorGregthis isnt your grandfathers pascal anymore
03:12.33dlynesThere's a reason why Asterisk isn't written in Pascal :)
03:12.57DoktorGregthere is also a reason web apps are not written in C:)
03:13.14DoktorGregway i see it
03:13.17dlynesDoktorGreg: Are you kidding?  Lots of webapps are written in C
03:13.32DoktorGregcompared to php?
03:13.40drfoomod2C#
03:13.54DoktorGreglol .net is .not,
03:13.54dlynesDoktorGreg: Lots of cgi apps (nowhere near as much as perl though), and most apache modules are probably written in C
03:14.22*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:14.24DoktorGregyah but the applications themselves
03:14.31DoktorGregare all in php and pearl
03:14.39drfoomod2DoktorGreg: why is that?
03:14.53DoktorGregbecause c is a systems language
03:15.07dlynesC is a language
03:15.11dlynesIt's not picky what it's used for
03:15.31DoktorGregit is not possible to write secure c end user applications, because of practicalty
03:15.40dlyneseh?
03:15.45dlynessince when?
03:16.00dlynesoh...nvm you said end user
03:16.02dlynesbut even then
03:16.09SplasPoodits totally possible
03:16.14SplasPoodit just might not be completely brainless
03:16.14dlynesIt's possible to write secure c end user applications
03:16.17dlynesYes, it is
03:16.20SplasPoodwhich is a big issue for a lot of developers..
03:16.24dlynesbut it's pointless writing end user applications in C
03:16.27dlynesIt's too much work
03:16.28DoktorGregdlynes, yas it is possible in theory
03:16.43DoktorGregas a practical matter, and in the real world of application development
03:16.54orlokADA!
03:17.05DoktorGregPROLOG!
03:17.10DoktorGregSCHEME!
03:17.27dlynespostscript!
03:17.31orlokhey
03:18.06orlokemacs.eps
03:18.38Nuggethttp://www.cafepress.com/nucleartacos.26746951  http://www.cafepress.com/nucleartacos.26229228
03:18.41DoktorGregVar'aq!
03:19.09DoktorGregVar'aq is the klingon programming language
03:19.32DoktorGregoh dear
03:19.47DoktorGreggoogle is reporting that there are several klingon programming languages
03:20.20drfoomod2DoktorGreg: you still haven't explain why it;s ".not"
03:20.23mmlj4isn't that militant programming? as opposed to procedural or object-oriented?
03:20.54DoktorGregbecause no one is using .net for anything serious
03:21.06DoktorGregexcept for microsoft
03:21.11dlynesDoktorGreg: Ummm...you must not be conscious
03:21.19DoktorGregbut even they seem to be scraping it in vista
03:21.32dlynesDoktorGreg: Either that, or you choose to believe your own reality
03:21.40drfoomod2DoktorGregL i disagree
03:22.12dlynesDoktorGreg: .net is used here in almost as many serious projects as J2EE
03:22.41drfoomod2dlynes: i assume you read the petshop white paper?
03:22.52dlynesYou pretty much can't get a job in this town without knowing .net
03:23.06dlynespetshop?  you mean the Sun sample J2EE app?
03:23.10Dorphalsigbrb
03:23.35drfoomod2and the .net version
03:23.38ledoktreDoktorGreg: Just wanted to let you know, I googled off your idea, and found a little 2 line deal, that I just tested, and seems to work good, for "Sip channel hunting" for outbound calls.  Just the one question left now :-)
03:23.49DoktorGregkewl
03:23.53dlynesNo idea...never read the white paper, either
03:24.16dlynesWhy do you ask?
03:24.26DoktorGregI avoid white papers
03:24.51DoktorGregI have found googling <technology> sucks to be more valuable assesment
03:24.51*** join/#asterisk Nugget (i=nugget@dazed.slacker.com)
03:24.55drfoomod2http://msdn.microsoft.com/vstudio/java/compare/
03:25.19dlynesDoktorGreg: You must not have too many pairs of clean underwear then
03:25.46DoktorGregI dont wear underwear
03:25.50dlynesewwww
03:26.34ledoktreAnyone know how to conditionally set CID for PSTN calls coming in? I.E., nothing set to PRIVATE CALLER, anything else, pass through?
03:26.51dlynesdrfoomod2: Yeah, it's probably all true
03:27.07dlynesdrfoomod2: After four or five years of J2EE, I've had my fill of business logic programming
03:27.18dlynesdrfoomod2: I avoid that type of programming like the plague now
03:27.44dlynesdrfoomod2: Besides, I can't keep up with all the bloody acronyms
03:27.46DoktorGregI never had to deal with the java buzz either...
03:27.52drfoomod2data tier programming for me is all templated
03:27.58dlynesdrfoomod2: You have to memorize like ten new acronyms every week
03:28.06drfoomod2ns
03:28.27drfoomod2like how java 1.2 was Java 2 but it took til 1.5 to get Java 5
03:28.32dlynesI swear J2EE is led by a bunch of engineers
03:28.37dlynesEngineers love acronyms
03:29.07justinu|laptophow about how solaris 2.5 turned into solaris 8?
03:29.31dlynesOr Slackware 5 turned into Slackware 7?
03:29.42justinu|laptopjava 1.5 is nice
03:29.54justinu|laptopJ2EE, otoh.... not so sure
03:30.07*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
03:31.09drfoomod2visual studio 2003 was version 7, but the latest is visual studio 8
03:33.28*** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net)
03:33.54*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
03:34.02DoktorGregthen a year or so after vista releases ms will release something else and break all the code again...
03:34.41DoktorGregI think that has been my main complaint with MS
03:34.50DoktorGregthey break all the code every couple of years
03:35.00litageis /usr/lib/asterisk/modules/codec_alaw.so used only for transcoding?
03:35.20*** join/#asterisk trbldwine (n=trbldwin@71.194.161.170)
03:36.11russellblitage: that is correct
03:36.18*** join/#asterisk Dorphalsig (n=asterisk@200.71.58.39)
03:36.27DorphalsigOk, now the weirdest things happen to me!
03:36.40DorphalsigI have everything green
03:36.42DoktorGregI look at MS performance on things like vista, and Live
03:36.54DoktorGregAnd letting IE die
03:36.58Dorphalsigbut I cant hear a thing
03:37.01Dorphalsignot even a dialtone!!
03:37.15drfoomod2?
03:37.15*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
03:37.18litagerussellb: does asterisk need a g729-equivalent of /usr/lib/asterisk/modules/codec_alaw.so to let g729 calls pass through it?
03:37.19DoktorGregthen selling the heck out of exchange....
03:37.28Dorphalsigyet I can make calls ... the phones ring, but nothing else happens
03:37.42SexyKenHey guys -- anyone here ever create queue status scripts in PHP using Manager API?
03:37.55russellblitage: no, g729 passthrough works without a codec translator.  however, you will need codec_g729 if you need to do transcoding
03:38.03DorphalsigSexyKen --> Nope, but it cant be that hard :P
03:38.11russellblitage: which you can get from Digium
03:38.27dlynesDorphalsig: Can you pick up the phoens when they ring and hear the other end?
03:38.37Dorphalsigdlynes -> Nope
03:38.47SexyKenDorphalsig:  Well -- I've got it working, to an extent.  Just not very good at organizing the data beyond queue info directly.  Trying to figure out how to display the data the right way...for instance....
03:38.58SexyKen...I can display queues, calls for each queue, hold time, etc.
03:39.15SexyKenBut I'm not good at figuring out how to show which agents are logged into which queues
03:39.19SexyKenAnd which calls belong to which queues
03:39.24dlynesDorphalsig: Maybe it's an issue with the configuration of the channel bank then, not asterisk?
03:39.47dlynesDorphalsig: or maybe it's how you've got span 3 and 4 configured?
03:40.03Dorphalsigits just the same way I had it working on my other * box
03:40.20litagerussellb: if i allow g729, ulaw, and alaw (in that order) for the [general] and [trunk] sections of sip.conf, and then try to make a ulaw or alaw call, asterisk tries to use g729
03:40.34SexyKenDorphalsig:  What is the issue you're having?
03:40.38dlynesDorphalsig: have you used those channel banks before then?  i.e. those specific ones, not same model, same brand, but same physical channel banks?
03:41.02blitzrageanyone ever put in ANSI color codes manually (like in a NoOp for example?)
03:41.02DorphalsigSexyKen, --> Cant hear a thing. Not even a dialtone. However I can ring other extensions
03:41.11blitzragerussellb: what are you still doing up! :)
03:41.19SexyKenAnd you've ruled out codecs?
03:41.34SexyKenIs this with a VOIP Connection or no?
03:41.39russellbblitzrage: writing a paper, heh
03:41.40X-Rob~centosbug
03:41.42jbotsomebody said centosbug was a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package.
03:41.48blitzragerussellb: ahhh -- fun stuff
03:41.49dlynesSexyKen: channel banks as in zaptel channels, not voip
03:41.52Dorphalsigdlynes --> Here also, with a previous asterisk. Just moved to a bigger server
03:42.04SexyKenGotcha.
03:42.16SexyKenAnd ringing other extensions still goes through the zaptel channels?
03:42.18russellblitage: um, i don't think i understand your problem
03:42.21dlynesSexyKen: He's using a te400p, with spans 1 and 2 going to pri's, and spans 3 and 4 going to channel banks
03:42.29SexyKenNo VOIP at all then?
03:42.33SexyKenGotcha.
03:42.34dlynesSexyKen: correct
03:42.50SexyKenHrmm.
03:43.07litagerussellb: the call is ulaw, but asterisk tries to use g729 rather than ulaw to talk with the device at the other end of the trunk
03:43.17blitzragerussellb: you don't happen to know how to control the color for some text on the console from the dialplan do you?
03:44.02dlynesDorphalsig: So what was causing the yellow alarms?
03:44.04*** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin)
03:44.08PakiPenguinmorning
03:44.14litagerussellb: however, if i remove g729 from the [general] and [trunk] sections within sip.conf, and then place a ulaw call, the ulaw call goes through fine
03:44.16Dorphalsigdlynes --> Dunno. Just changed the card to another slot
03:44.17russellbblitzrage: don't think you can, or maybe you can put the color escape codes in your text
03:44.23dlynesDorphalsig: ah...weird
03:44.27russellblitage: yeah, that's expected behavior
03:44.35litagerussellb: why is that?
03:44.39blitzragerussellb: yah -- just looking for the escape codes now -- I forgot the format
03:44.55russellblitage: so, what you could do is create two peers for the trunk.  one that allows g729 and one that does now
03:45.08Dorphalsigdlynes, --> yeah, that0s why my greeting was "the weirdest things happen to me"
03:45.14russellblitage: then, in the dialplan, check the SIPCODEC variable to see if the call was g729
03:45.19russellband use the appropriate peer
03:45.29dlynesheh
03:45.37dlynesSame here
03:45.42litagerussellb: thanks, i'll check that out. i'm curious as to why this happens though
03:45.42russellblitage: excuse me, the SIP_CODEC variable
03:45.58dlynesYou wouldn't believe some of the weird ass shit I've experienced with asterisk installs in rural areas
03:46.29litagerussellb: or why this is "expected behaviour"
03:46.37Dorphalsigdlynes --> thanks for the rural
03:46.39NuggetI didn't know asterisk could run party lines.  ;)
03:46.41russellblitage: because you told asterisk that when making calls to this peer, you prefer g729
03:47.15litagerussellb: but i specified that g729, ulaw, and alaw are allowed
03:47.25mmlj4dlynes: wanna give some examples of weird?
03:47.32dlynesNugget: I don't think you can even still get party lines here
03:47.34russellblitage: and when it makes that call, it has no knowlege of the other side of the call
03:48.02russellblitage: the called channel is not bridged to the calling channel at this point
03:48.07mmlj4nuts... i really wish I could land a job as an * tech, instead of desktop linux support
03:48.09dlynesThat's the first time I've heard anyone even mention party lines in the last probably fifteen years
03:48.32dlynesmmlj4: like?
03:48.48mmlj4<dlynes> You wouldn't believe some of the weird ass shit I've experienced with asterisk installs in rural areas
03:48.51russellblitage: so there is no way for chan_sip to say "hm, let me go find out what i'm going to be bridged to in the future and change my format preferences accordingly"
03:49.05litagerussellb: okay, that makes sense
03:49.08dlynesmmlj4: I had an entire asterisk system jsut cease to function any longer; i couldn't even log into it anymore
03:49.12litagerussellb: however...
03:49.20dlynesmmlj4: freaking huge emf fields in the telephone closet
03:49.25mmlj4heh
03:49.47dlynesbut for whatever reason the old nortel keysystem just chugged along like the emf fields didn't even exist
03:49.58mmlj4shielding?
03:50.14dlynesif there was any, it's built into the keysystem (cics)
03:50.17mmlj4it's not like PC cases are any good except for supporting the cupholder
03:50.18litagerussellb: in [general] and [trunk], i allow ulaw and alaw (in that order). i then placed an alaw call, and the call went through fine
03:50.21drrayold pbx's are a lot more robust than a digium card in a beige box
03:50.52russellblitage: because asterisk knows how to translate between alaw and ulaw
03:50.55dlynesWe ended up giving up on ever getting asterisk to work there, and sold them a panasonic instead
03:50.57DorphalsigOk its up and running
03:51.11Dorphalsigjust needed to change coding for the channelbanks and Boom! They work!
03:51.18dlynesDorphalsig: aha!
03:51.18litagerussellb: if asterisk is translating between ulaw and alaw, what sort of message on the asterisk console should i expect?
03:51.19drrayyay!
03:51.24drrayBoom!
03:51.29russellblitage: you won't see anything about that, probably
03:51.38litagerussellb: how come?
03:52.11dlyneslitage: unless you've got debugging turned on, you won't see anything about it, unless there's a problem
03:52.15*** join/#asterisk bmg505 (n=leon@dsl-165-130-138.telkomadsl.co.za)
03:52.26russellbthat's just how asterisk is supposed to work
03:52.29*** join/#asterisk Curi (n=la@pc-38-169-104-200.cm.vtr.net)
03:52.41Curihello
03:53.00Curiany one knows if there's a way to handle the ANSWEREDTIME variable?
03:53.52LostFrogShouldn't echo use just as much bandwidth as a normal conversation?
03:55.16litagethanks russellb and dlynes, what you've just told me has been REALLY helpful  :)
03:55.34Math`http://pastebin.ca/50339
03:55.46Math`trying to plug a meridian pbx with a TE110P
03:55.59Math`(this is what happens when I dial from an M1 phone)
03:57.13russellblitage: i appologize but I just looked and SIP_CODEC isn't for what I told you
03:58.12russellblitage: SIP_CODEC is a variable you can *set* so that when a call is made to a sip channel, it will attempt to use that format
03:59.06russellbin any case, i hope i've helped out the understanding of how it works ...
03:59.55dlynesrussellb: When you set that variable, will it force the sip phone that's calling on that channel to use that codec instead?
04:00.55russellbno, the way you would use it, is execute Set(__SIP_CODEC=ulaw) in your dialplan.  The "__" indicates that you want this variable to be inherited into any new channels (such as the channel created when you run Dial)
04:01.08russellbthen, the *new* channel, the one created by Dial, will try to use that codec
04:01.33dlynesbut the existing one won't use it, unless it's already using it?
04:01.46russellbit has no effect on the current channel
04:02.03dlynesthe reason i'm asking is because i thought it might help mitigate the buggy codec negotiation in asterisk
04:03.15*** join/#asterisk viLeR (i=1000@66.128.47.232)
04:04.07russellbok, wow, i'm not doing well tonight.
04:04.25russellbI retract everything I've said about SIP_CODEC so far.  (You can tell I'm not a SIP person).
04:04.44dlyneslol
04:05.00russellbso, Set(SIP_CODEC=whatever) in the extension handling an *incoming* SIP call *before* the channel is answered
04:05.17russellband Asterisk will try to negotiate that codec with the incoming call
04:06.05russellbsheesh.
04:06.13dlynesah...cool
04:06.16dlynes?
04:06.20russellbi better stop before I give out some more bogus info
04:06.29Snake-Eyeshehe
04:06.35dlynesdon't worry about it
04:06.44OliverXrussellb: i call with my handy to my home -> isdn symste -> isdn fritz card -> asterisk -> x-lite
04:06.46russellbat least i corrected myself :)
04:06.48dlynesYou're one of the most knowledgable people i've seen on this channel
04:06.48OliverXi have a connectionen
04:06.54OliverXbut no sound transfer
04:06.56drrayand you are willing to help
04:06.58OliverXhave you an idea?
04:07.10dlynesyeah...what drray said
04:07.26russellbwell thanks :)
04:07.32dlyneswillingness to help is the most important...even if you give out erroneous information, it often helps lead the asker on the right path
04:07.34russellbI usually just lurk in here ;)
04:07.41OliverXHm
04:07.49OliverXi cant find the mistake :(
04:08.00OliverXi can start and hangup  the call
04:08.05OliverXbut no sound :(
04:08.32Snake-Eyesrussellb, is there a way to get the codec of the incoming call?
04:09.09russellbSnake-Eyes: there will be in 1.4 :)
04:09.30russellbannnnnd, it's not channel specific
04:09.38OliverXrussellb ;)?
04:09.54dlynesOliverX: not specific to sip/mgcp/sccp/h323/...
04:10.07OliverXhm?
04:10.26dlynesOliverX: it'll work on all voip channels
04:10.32russellbSnake-Eyes: Set(CODEC=CHANNEL(audioreadformat))
04:10.38OliverXyes but i have no sound
04:10.48OliverXthe call is open
04:11.00russellbthat will set the variable CODEC to be the channel's "read format", which is the format that audio is "read" from the channel
04:11.12OliverXbut if i talk in my handy, i have no sound on my x-lite in my local lan
04:11.17dlynesOliverX: I think we were both thinking you were talking russel's reply to snake-eyes
04:11.25russellbbut ... that's brand new, not in 1.2
04:11.38Snake-Eyesah ok
04:11.52*** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com)
04:12.05OliverXdlynes: what had he write?
04:12.12tmccraryWhat would you say is the best SIP IP phone under $100?
04:12.18dlynesSnake-Eyes russellb, is there a way to get the codec of the incoming call?
04:12.18dlynesrussellb Snake-Eyes: there will be in 1.4 :)
04:12.21Snake-Eyes1.4 6+ months away isn't it?
04:12.22tmccraryNo religious wars please :)
04:12.30dlynesSnake-Eyes: June iirc
04:12.32russellbSnake-Eyes: no, about 2 months
04:12.43Snake-Eyescool
04:13.09dlynestmccrary: I didn't know you could get a good one under $100
04:13.42dlynestmccrary: The only one I can think of that might be close is the Grandstream Budgetone 101/102
04:13.57dlynesactually...come to think of it...it's under $100
04:14.02tmccraryI'm not saying "good" but what one do you think is best at that price range. Obviously, these are budget phones.
04:14.05dlynesand it's relatively good
04:14.06russellbalright, nice talking to you guys, i really need to go write this paper :)
04:14.17dlynesIt's just damned ugly
04:14.27tmccraryYeah, have you used the ArtDio phones?
04:14.37dlynesnever heard of them
04:15.00dlynesalso the Aastra guest phone (single line phone) is about $110 Cdn
04:15.13dlynesThe aastras are damned nice phones
04:15.29tmccraryI wonder what the cheapest PoE phones run
04:15.31dlynesThey pretty much look like the newer style nortel phones
04:15.40dlynestmccrary: That would probably be the Aastras
04:15.50dlynestmccrary: Or maybe the Sipura 941's
04:16.15tmccraryThe Aastra 9112i is a pretty high quality phone?
04:16.27dlynesYeah...not quite as good as polycom, but pretty close
04:16.50dlynesWe've got a bunch of the 9133i's deployed
04:17.12dlynesmuch better than any of the other crap ip phones we've tried
04:17.14tmccraryyeah, these look pretty nice
04:17.25dlynesSuper nice speaker phone too
04:17.31Snake-EyesI have spa941 next to me, I hate the polycom we have in the office
04:17.33dlynesIt's full duplex even on the entry level phone
04:17.55dlynespolycoms don't become full duplex speaker phones until you get up to the 501
04:20.10tmccrarythe 480i's look nice too
04:20.29dlynesYeah, but the 480i's are a hell of a lot more than $100
04:20.31tmccraryWhere did you get these for $110 (presumably american currency?)
04:20.38dlynes$110 Cdn
04:20.48tainted-that's like 4.85usd
04:20.50dlynesI think that's what it was...
04:21.10dlynes$110 Cdn is about $95USD
04:21.19tainted-man i used to go up to vancouver and live like a king
04:21.29tmccraryWhere do you shop for these dlynes?
04:21.30mishehutonight is not my night.  for some reason I'm getting a runaway process when I make an std::string object then getline(cin,string);.
04:21.51tmccraryC++ makes baby jesus cry
04:21.52tmccrary:((
04:21.53dlynestmccrary: Are you going to be buying wholesale, or retail?
04:22.39tmccraryWell, small quantities, so retail I suppose. Unless there's some cool wholesale outfit online somewhere
04:23.10dlynestmccrary: Well, if you go wholesale, Williams Telecom has pretty good prices on Aastra phones
04:23.15mishehutmccrary: shut that damn baby up
04:23.17mishehuheh
04:23.52tmccraryI'm j/king. I code primarily Java EE apps, so I make baby jesus cry blood from his eye sockets
04:24.48dlynestmccrary: http://www.williamsglobal.com/products.asp?type=Show&catId=20574&manId=114
04:25.05dlynestmccrary: I think voipdepot.ca and voipsupply.com also sell them
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04:25.34dlynestmccrary: Williams, Voipdepot are both Canadian companies; Voipsupply is an American company
04:25.46tmccrarywow I found one for $195 retail
04:25.50tmccraryw/ PoE injector too
04:25.55tmccraryhttp://www.neobits.com/do/dtls?pid=9734
04:26.02dlynesWilliams is wholesale; Voipdepot and Voipsupply are both retail
04:26.10tmccraryoh wait, no injector
04:26.24tmccraryCool, thanks for the links
04:26.29dlynesThe injector's about $45Cdn, if I remember correctly
04:26.36dlynesagain, that's wholesale pricing
04:26.38tmccraryI actually checked VOIPSupply earlier, that's where I usually buy stuff
04:27.40dlynesYeah...Williams is the best pricing I've found anywhere for Aastra equipment
04:27.56dlynesThey were even better than the pricing the manufacturer was willing to give me
04:29.07tmccraryDo they have a minimum quantity?
04:29.20tmccraryI only need like 8 phones
04:29.54dlynesNo minimum quantities, no
04:30.14tmccrarycool
04:31.05dlynesJust talk to Don Wright there
04:31.16dlynesHe's one of the sales guys in charge of wholesale accounts
04:31.29dlynesYou'll need to set up an account with them to get wholesale rates
04:31.40tmccrarydo they sell to the US?
04:31.44dlynes1-800-982-3333
04:31.45dlynesProbably
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04:32.28tmccrarycool, thanks
04:33.34dlynesAnyways...they've been around for over 20 years, so they're quite reputable
04:33.50dlynesMost of their stuff is quite overpriced as far as wholesale goes, but their Aastra stuff is quite cheap
04:33.58dlynesAnd they also sell their own Asterisk PBX system
04:34.31OliverXhttp://rafb.net/paste/results/XomhWU49.html have anyone here an idea(i have only a connectionen between the handy and xlite, but no sound
04:37.41tmccrarydlynes, do you work for williams?
04:37.50tmccrary;)
04:38.33dlynesnope
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04:38.38dlynesThey're jsut one of our suppliers
04:38.44dlynesThey're in Toronto
04:38.47dlynesWe're in Vancouver
04:40.26dlynesanyways...back to coding
04:40.26dlyneslaters
04:42.38tmccrarylater aaand thanks again
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04:59.11drrayok, how is Voip pronounced?  are people really calling it "voyp"
04:59.12drray?
04:59.51distortionpretty much
05:00.07drrayso I can't kill my cube-mate
05:00.09drraygreat
05:00.45justinu|laptophaha
05:00.46justinu|laptopn00b
05:00.55distortionhehehe
05:01.02drrayI just heard someone call it that
05:01.13drrayand wanted to double check before I beat him with a channel bank
05:01.36distortionput your channel bank down, i have a deal for you ;)
05:01.43drrayit's a zplex
05:01.46drrayyou don't want it
05:02.16drrayplus he's pretty mushy, I doubt it would suffer much
05:07.12tainted-are people really saying voyp?
05:07.55distortionits voyeep, gosh!
05:07.58tainted-they should be shot
05:08.16tainted-gooyee
05:08.48tainted-but i guess it's better than 'vee-o-i-pee'
05:09.01drrayI call it voice-over-ip
05:09.05drraybut I like to talk
05:09.07dlyneswow
05:09.15distortion...
05:09.17dlynes11 acre lot for $42K
05:09.25drrayin pakistan?
05:09.32dlynesThunder Bay :)
05:09.41tainted-call it voice delivered via packets over internet protocol then
05:09.53dlynesI don't even think I could get enough land to build an outhouse on here for $42K
05:10.06tainted-vdvpoip
05:10.31distortionyou went too far tainted.
05:11.11tainted-that's a good call
05:12.03drraywell, I guess I'm not the only one ey-ar-kay'd by hearing it called voyeep
05:13.29tainted-i can handle hearing 'Sequel' when people refer to SQL.. maybe b/c it's an existing word
05:13.37distortioneek
05:13.46distortioni cant stand when people say sequel
05:13.52drrayme either
05:13.58tainted-even dbas say sequel
05:14.21drraywhen I was a yute I referred to ASCII as asc-2
05:14.28tainted-i hate hearing fully grown bearded men say 'IAX' as 'eeks'
05:14.39tainted-i want to smack the brokebackness out of them
05:14.40distortionnot my russian dba, he says- S-Q-L-L-L
05:14.41drrayI was giving a presentation at the computer club
05:15.46drrayin soviet russia, SQL says you.
05:15.50distortiondrray, you should not be allowed to present at a computer lab if you dont know how to pronouce voip
05:15.50drrayok
05:15.58drrayI was 12
05:25.06DelmarIf anyone on here has John at VoipJet's MSN contact in their list, or if he is in here, please private message me. thanks.
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05:39.40Mikeanyone knows of a good way to make loadbalancing with asterisk?
05:40.01dlynesMike: Did you check the high availability section of the wiki?
05:40.36Mikedlynes, not really i checked on ser and on sip load balancer but i will now go to that seccion
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05:41.30dlynesMike: Yeah...there's a whole section there for HA, load balancing and a few other things, including carrier grade linux
05:44.13dlynesHas anyone used the 'w' for the dial string on zap channels?
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05:44.18dlynesi.e. the pause character?
05:44.20asterboyquit
05:44.26asterboyoops
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05:47.05PBXtechanyone know this error: file.c:1032 ast_waitstream: Unexpected control subclass '-1'
05:47.31terrapenhrmmm...i think i made my T1 crossover cable incorrectly
05:48.08Mikeultramonkey seems like a solution
05:48.16terrapenultramonkey?
05:48.21terrapenoh, the HA
05:48.35Mikeim going to use it for load balancing
05:48.44terrapenfor SER?
05:48.50Mikei have to router calls in 15 asterisk servers
05:49.06terrapenwhy not put SER in front of Asterisk, mike
05:49.08argos73verification question - it's possible to transfer an existing call (SIP<>Zap) into a meetme conference, right?  (I'm unable to test from where I am0
05:49.08Mikeno ultramonkey + 15 asterisk servers
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05:49.26PoWeRKiLLhi
05:49.30terrapenunless, of course, you have lots and lots of PRIs
05:49.32Miketerrapen, i dont know never used ser before and seems like i have to learn how to use condicionals
05:50.15terrapenmike, if you're just trying to get a lot of SIP clients running smoothly with Asterisk, SER might be a much better choice
05:50.31terrapenits more efficient at the simple SIP<--->SIP stuff
05:50.32PoWeRKiLLI have different * box with different E1 on them I need to send an hangupcause to the PRI so the call will not be answered by the * box but will be send to another E1 on another box someone know wich hangupcause do I have to send ?
05:50.40Miketerrapen, do you have any webpage with a dummy howto for non programmers?
05:50.57terrapenand can send calls to your * servers when the client needs something like voicemail, etc
05:50.58Miketerrapen, all the docs of ser i have read involve condicionals and condicionals and more condicionals
05:51.36terrapenhttp://www.voip-info.org/wiki/view/SIP+Express+Router
05:51.45terrapenalso, do some googling for SER / Asterisk architectures
05:51.58Mikeok
05:52.08terrapenyou don't have to be a programmer but if you are going to admin an installation large enough to require 15 * servers, you better learn to configure these
05:53.14terrapenhttp://www.voip-info.org/wiki-Asterisk+at+large
05:53.16terrapenthere you go
05:55.06Mikeim reading that now
05:55.32Mikeif (uri=~"^sip:0[0-9]*@yourdomain") {
05:55.39Mikethats the stuff i cant really figure out
05:55.49terrapenthat's a simple regular expresion
05:55.53terrapenlike in Perl
05:56.40terrapenit says, "If uri starts with sip:0, followed by zero or more digits in the range 0-9, followed by "@yourdomain"
05:57.04Mikewhat does the ~ and the ^ mean
05:57.04Mike?
05:57.30terrapenuri =~ ....  means that a regular expression is going to follow
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05:57.47terrapenif the regular expression matches some text, it will return true
05:57.50Mikecan i do ser without going on regular expresions?
05:58.00terrapenand the block after the 'if' statement will run
05:58.13Mikeif not ill google a good regular expresions doc
05:58.20terrapenregular expressions are easy to learn and a good tool for any system admin
05:58.27terrapenthey look scarier than they are
05:58.46terrapenonce you learn them, you can do very powerful things
05:59.00drraylearn them when you have a need
05:59.01terrapenhttp://en.wikipedia.org/wiki/Regular_expression
05:59.09drraydon't bother just to learn them
05:59.29terrapenwell, it would help him with SER
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05:59.33Mikelearning regular expresions + ser + testing load balancing will be a fun trip :)
05:59.40terrapenand it's a good thing to have for future employment prospects
05:59.40Mikelet me check the wikipedia on that
06:00.02diLLecto fully learn and understand regular expressions needs a lot of time.
06:00.03terrapenSER machines are easily load-balanced
06:00.16terrapendillec, yes, but it only takes a few minutes to get the basics
06:00.23diLLecthats correct
06:00.35terrapenie, * +  [ ]  ^   $
06:00.39terrapenand .
06:00.42asterboy~echo
06:00.47jbotecho is, like, normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. echo, echo  ..echo... cho..o... ..  It also helps ADHD people keep track of their conversations. It's all good!
06:01.03terrapenif you can learn those six things, you are 90% there
06:01.54terrapeni think its my wacky homegrown T1 crossover cable
06:02.00terrapennot getting any link lights
06:02.37Miketerrapen, if wikipedia enought?
06:02.41terrapenyeah
06:02.48terrapenit explains them
06:02.55terrapenbut you may find a simpler document somewhere
06:03.39Mikeok lets google
06:03.50Mikeperl regular expressions right?
06:03.56terrapenyep, those will work
06:04.03dlynesWould anyone happen to know where in chan_zap.c I would need to go, to modify the length of the wait in the dial string?
06:04.23terrapentime to go home
06:04.28terrapenive been at the office for 14 hours
06:04.35argos73there's also an o'reilly book on regexps...  a little pricey ($40) if you're not "dedicated to the cause", but a good book..
06:04.42Mikebye terrapen
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06:06.57dlynesargos73: what do you mean by 'dedicated to the cause'?
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06:08.21argos73hehe - there's a lot of good docs on regexps out there, but having a book (single go-to source) is nice at times
06:08.51dlynesargos73: definitely...i prefer books myself, to online documentation...I just don't understand what you meant by that expression
06:09.36argos73just meant that for Mike, who's just trying to the the basics, it might be a little much at this point
06:09.51argos73unless he really likes books
06:09.56argos73:)
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06:10.09dlynesah
06:10.22dlynesbesides the thing i like about o'reilly books
06:10.49dlynesthey often divert a certain percentage of the proceeds of any book on open source software to the project it documents
06:11.17dlynesso it's a good way to indirectly support the developers of your favorite software
06:11.26argos73true...  I do have a decent collection of them myself.
06:11.58dlynesyeah...o'reilly has some of the best books around, too
06:12.21dlynesout of all my favorite books, I'd say probably 95% of them are o'reilly titles
06:12.28argos73even "upgraded" my old Sendmail book to the 2nd release version...
06:12.36orlokyeah, they make some of the classics
06:12.44orlokllama book, camel book, etc
06:12.54orlokbat book, etc
06:13.09coppicethree toed sloth book
06:13.31coppicewhy the choice of such oddball animals?
06:13.44dlynesyeah...those pocket references are pretty cool, too
06:14.02coppiceDo their python books actually have pythons on them?
06:14.09dlynesyes
06:14.40coppiceI expected they might have, say, a rabbit and claim its the prey of the python :-)
06:14.42dlynesand their asterisk book really has an asterisk on it (starfish)
06:14.48argos73some of the animal selections are appropriate...  others are "gee, this particular species just doesn't get enough press"
06:15.53dlynesyeah...like what the hell does a toucan have to do with qt?
06:16.11argos73only a bird-brain would use QT?
06:16.11argos73:)
06:16.24dlyneslol...no....qt is better than gtk, i think
06:16.40coppicetwo can play at the GUI game?
06:16.50dlynesThe animal on the cjkv book though is sorta appropriate
06:16.55dlynesIt's a blowfish
06:17.07argos73damn these upper-level libraries - everyone should be required to do direct X calls
06:17.32coppicewhy is a blowfish appropriate?
06:17.45florz"X calls"? You mean read() and write(), don't you?
06:18.09dlynescoppice: it's a popular fish for sushi with the japanese :)
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06:18.25dlynesthat's why i said sorta appropriate
06:18.43argos73XMoveCursorOverHereAndMakeItDoNothing()
06:18.56coppiceso how is that appropriate to 90+% of the population covered by CJKV?
06:19.05dlynesIt isn't
06:19.11dlynesi said 'sorta'
06:19.20dlynesI didn't say exactly
06:19.22dlynesI didn't say mostly
06:19.24dlynesI said sorta
06:19.36argos73just like "good software sorta comes out of Redmond"
06:19.46coppicesorta parochial if you ask me
06:19.57dlynesbesides
06:20.09dlynesdon't other asian cultures also eat the blowfish?  Just not in sushi?
06:20.29dlynesI would think it's also eaten in Chinese coastal cities...
06:20.30coppicenope. only when eating japanese food
06:21.18dlynesah...so it's not even eaten in hong kong, eh?
06:21.21coppicemost chinese won't even eat it then, because of the health risk
06:21.51argos73hmmm..  maybe I should serve blowfish at the next company cookout.  might weed out some of the undesirables...  :)
06:22.00dlyneslol
06:22.33dlynesThe japanese that I know that do eat it, eat it because of the rush it gives them because of the chance that it might give them paralysis
06:23.09dlynesto me, that'd be a reason not to eat it
06:23.11coppicewell, its always the residual poison that attracts people. it makes your mouth tingle.
06:26.17dlynesso do people actually eat all that weird ass stuff in hong kong, too?
06:26.28FuriousGeorgedlynes you're here a lot lately
06:26.30dlyneslike live monkey brain and bull penis?
06:26.48FuriousGeorgewhats wierd about monkey brain and bull penis
06:26.59FuriousGeorge?
06:27.07coppicelive monkey brain, dog, and various other things are against the law. deer penis is a lot more common than bull penis
06:27.37dlynesor is that live monkey brain mostly an old wive's tale?
06:27.50FuriousGeorgei saw it on faces of death
06:28.01FuriousGeorgethey stick the little guy under the table
06:28.10coppicei've never seen anyone eat it, but I gather they do
06:28.25FuriousGeorgeand there's a whole in the middle, that his head can fit through
06:28.49FuriousGeorgeand they kinda close him in there and give the dudes at the table a hammer
06:29.34FuriousGeorgethats how my cousin died
06:30.01Math`ok
06:30.14Math`If I fedex you a can of soup will you setup an enterprise pbx?
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06:30.32FuriousGeorgedepends on the soup spoon
06:30.42Math`a Knor(tm) one
06:30.50dlynescoppice: well, i've seen the bull penis served up in some famous restaurant in taibei...it was on some morning taiwan talk tv show
06:31.04FuriousGeorgethen no
06:31.38coppicedlynes: we have bull penis, but like I said deer penis is the real thing to eat
06:31.42Math`anybody compiled chan_oh323 successfully on 1.2 (so chan_oh323 version 0.7.3
06:32.01dlynescoppice: even then, most of the people that tried it, said that was enough for them...they've tried, and won't bother sampling it again :)
06:32.22FuriousGeorgeshoot we eat that here in the states.  its usually a main course on the reality TV shows.  and since everyone seems to be on one of those these days...
06:32.43coppicedlynes: it would be classified as "chinese medicine". you aren't expected to actually enjoy medicine :-)
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06:33.05dlynescoppice: yeah..same thing with the bull penis...it's considered to be an aphrodisiac :)
06:33.23tainted-well u do have a penis in your mouth ...
06:33.36tainted-i guess that could conjure up sexual innuendo
06:33.44coppicesounds vaguely gay :-\
06:34.27coppiceactually you don't usually eat the penis. you make a broth from it, and drink the broth
06:36.13Az_auany advice on killing a minor echo on a te110p when making outbound calls.. only the caller can hear their own echo.. other party does not make an echo nor can hear any echo
06:37.17dlynesAz_au: is it on all recipients, or just particular recipients?
06:37.29tainted-Az_au have u tried bull penis broth.. cancels echo for me pretty well
06:37.38Az_audlynes: appears to be all but is only monir.. it is amplified when using a headset
06:37.40dlyneslol
06:37.56dlynesAz_au: That's why I had asked you if it was particular phones or not
06:38.07dlynesAz_au: Some phones happen to have bad feedback on them
06:38.19dlynesAz_au: Which woudl cause echo on the remote end, but not the local end
06:38.45Az_audlynes: i've tried it on a softphone as well and it seems to be there as well and the echo does not happen on a local extension call
06:38.50Math`FuriousGeorge: what about a box of Kraft Dinner?
06:39.13Az_auwhich is why i think it's a setting on the te110p side of things
06:39.15dlynesAz_au: yeah...no analog on local...that's why :)
06:39.29dlynesAz_au: Which echo canceller are you using?
06:39.38dlynesAz_au: MG2 is the latest and greatest
06:39.54argos73Macaroni and Bull Penis?
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06:40.38Az_audlynes: i'll have a look into it.. ta
06:44.15DoktorGregthe only echo i have encoutered is on computer sip clients where the speaker is feeding back into the mic
06:45.14coppicecomputer based acoustic echo cancelling doesn't work properly with a lot of sound cards. its a hardware issue
06:45.17dlynesDoktorGreg: Do you use analog lines?
06:45.29DoktorGregno, pri and voip only
06:46.04DoktorGregbut i dont recall echo on my test rig either, with a TDM400
06:46.38DoktorGregbut that was all on lan...
06:47.43DoktorGregwhere does the echo come from on analog lines?
06:48.27coppicethe echo fairy
06:48.28RGi_Morn ppls :)
06:48.42DoktorGregspeaking of which
06:48.54DoktorGregI have a spare tdm400p with 2 fxo ports
06:49.38Az_audlynes: that worked fine, thanks
06:49.43coppicethe echo comes from every hybrid down the line, whether the line is analogue or digital
06:49.46dlynesAz_au: The mg2?
06:49.50Az_audlynes: yep
06:49.55dlynesAz_au: cool
06:51.00dlynesAz_au: I've been using that with libpri-trunk/zaptel-trunk now for about 2 weeks, and i haven't had a problem with echo
06:51.12dlynesAz_au: mg2 with zaptel-trunk is the default echo canceller now
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06:52.02Az_audlynes: ya mine was set to KB1
06:52.46coppiceall the cancellers in * are too crude to be robust. whether you see problems is more luck than design
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07:12.27ccedhi all~
07:12.40Nuggetmoo.
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07:28.48RoyKrotfl. danish railways had a stop, so they decided to put the passengers in a taxi to get them home, from copenhagen to Narvik in nothern norway, shortest way between the two is >2000km
07:29.30tparcinaHinting, Cisco, SIP, has anybody make it working?
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07:33.51coppiceI just added the most frequent question I get about spandsp to the FAQ - http://www.soft-switch.org/spandsp_faq/ar01s12.html :-)
07:35.01creadurexhehe
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07:41.39*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
07:42.28dlynescoppice: duh? :)
07:45.51ccedgood
07:46.10*** join/#asterisk stoffell (n=stoffell@147.63-136-217.adsl.skynet.be)
07:46.13*** join/#asterisk Eggplant (i=No@dsl-731.cascadeaccess.com)
07:48.56*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
07:49.50SexyKenIs live monitoring possible in asterisk?
07:50.18coppicelooking for ceased users? :-\
07:50.33coppices/ceased/deceased
07:53.53creadurexSexyKen: define live monitoring? audio ?
07:54.09SexyKenYea - for instance..we run a call center.
07:54.16SexyKenWe'd like to monitor certain techs for qa
07:54.23SexyKenListen live
07:55.11creadurexif im not completely wrong that should be possible as a manager
07:55.16creadurexto eavesdrop etc
07:55.28SexyKen"as a manager"
07:55.31SexyKen?
07:55.43coppiceor as a secret agent for a foreign power, maybe :-)
07:56.04creadurexyeah or if you use AT&T for hosting, anyone can listen ;)
07:56.11SexyKenHrm.
07:56.16SexyKenI am confused now
07:56.21*** join/#asterisk n4y (n=frodo7@host-ip237-209.crowley.pl)
07:56.47creadurexfear not
07:56.48creadurexhttp://www.voip-info.org/wiki/view/Asterisk+record+calls
07:56.54creadurextheres some info there
07:56.57coppicethere are spying facilities, with obscure names that confusingly include "spy"
07:57.26n4yanyone can halp me? when I am trying to call via 4 E1 card I get: "No d channel is available"
07:58.06coppicemaybe should should make time slot 16 on each E1 a D channel
07:58.20n4yI did it
07:58.22*** join/#asterisk hads (n=hads@203.109.245.87)
07:58.38coppiceapparently not, or it wouldn't complain
07:58.50n4ydchanel = 16
07:59.24n4yin zapata.conf
07:59.34n4yany other suggestion?
07:59.35coppicefor each E1?
07:59.43n4yonly for one
07:59.52*** join/#asterisk shanky (i=jramirez@217.11.114.145)
07:59.52n4ysince I use only one
08:00.06n4yonly first E1 is used
08:00.48creadurexdumb question here.. is the astdb cached? I want to force all SIP peers in the astdb to re-register, a reload didnt help
08:01.39*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
08:01.51nextimen4y : dchanel = 16 in zapata.conf?
08:02.04coppicen4y: did you allocate the d channel in zaptel.conf?
08:02.06SexyKenWhat's the proper way to start asterisk while specifying a conf
08:02.08nextimeare you sure of what you say?
08:03.00nextimen4y : you need to specify the dchannel in "zaptel.conf", not in zapata.conf, and dchanel is wrong, you need something like:
08:03.00nextimespan=1,1,0,ccs,hdb3
08:03.00nextimebchan=1-15,17-31
08:03.00nextimedchan=16
08:03.08n4ycoppice: sorry I was meaning zaptel.conf
08:03.30n4ynextime: yes i did it
08:04.11coppicen4y: no you didn't. in zaptel.conf you need what nextime said, not what you said. in zapata.conf you need dchannel=16
08:04.56n4ycoppice: yes i did it in zaptel.conf
08:05.17nextimecoppice : are you sure about dchannel=16 in zapata.conf?
08:05.41coppicen4y: make up your mind. you said you put dchannel=6 in zaptel.conf. that's wrong. its dchan=16 in zaptel.conf
08:05.44nextimei don't need it in my * boxes
08:06.21n4yi get: "No d channels available. Using Primary channel 16 as D-channel anyway!"
08:06.27coppicedunno. maybe it defaults these days. it used to be required
08:07.09n4ysorry of course I set : dchan =16
08:07.59SexyKenGuys
08:08.00SexyKen[res_config_mysql.so]Apr 21 02:04:28 WARNING[1903] loader.c: /usr/lib/libmysqlclient.so.15: version `MYSQL_5.0' not found (required by /usr/lib/asterisk/modules/res_config_mysql.so)
08:08.03SexyKenI'mg etting this now
08:08.19coppicen4y: set dchannel=16 in zapata.conf, and I think that message should go away. i think its just telling you it is defaulting to 16 because you didn't specify a d channel
08:08.44n4ycoppice: i will try
08:11.40*** join/#asterisk RoyK (n=roy@213.160.242.134)
08:13.09fourcheezeSexyKen: do you have mysql 5 client libs installed?
08:13.17SexyKenNot sure.
08:13.18SexyKenBut
08:13.20SexyKenI dont have mysql5
08:13.22SexyKenNever did
08:13.46fourcheezewhere did you get your asterisk ?
08:14.08SexyKenHad someone install it.
08:14.19SexyKenI updated mysql client whatever to 15off
08:14.21SexyKenfrom 15
08:14.24SexyKenFor another app
08:14.25SexyKenthat required it
08:14.27SexyKenAnd then got that
08:14.48fourcheezewhich distro/OS ?
08:14.54SexyKenDebian
08:14.55SexyKenLinxu
08:15.04fourcheezetry installing the mysql 5 libs
08:15.08*** join/#asterisk razu (n=razu@tln-kontor.norby.ee)
08:15.17SexyKenWell
08:15.23SexyKenlibmysqlclient15off is installed
08:15.39fourcheezemysql-client-5.0  should do it
08:16.01fourcheezedebian should be able to cope with both at once
08:16.09fourcheezesee what it tries to do when you install 5.0
08:16.11SexyKenWell - it is saying to remove it
08:16.14SexyKenthe other ersion
08:16.16fourcheezeok
08:16.26fourcheezeso now you know how your 5.0 went missing
08:16.35SexyKenBut
08:16.35n4ycoppice: i get still the same
08:16.37SexyKenI never had it.
08:16.54fourcheezeare you sure the person who installed * didn't install 5.0?
08:17.06SexyKen100%
08:17.11SexyKenThe database is in 4.x
08:17.18fourcheezethat doesn't matter
08:17.25fourcheezewe're talking about the client libs
08:17.30SexyKenRight - I messed up
08:17.33SexyKenI replaced the 15 libs
08:17.34SexyKenwith:
08:17.34SexyKen/var/cache/apt/archives/libmysqlclient15off_5.0.20-1_i386.deb
08:17.40SexyKenI guess that's ad.
08:17.41SexyKen*bad
08:17.53fourcheezenot sure what the off means
08:18.13fourcheezebut those are 5.0 libs
08:18.20SexyKenBut I think I had this before:
08:18.20SexyKen/var/cache/apt/archives/libmysqlclient15_5.0.16-1_i386.deb
08:18.23dlynesfourcheeze: probably 'official'
08:18.38fourcheezeI'm guessing they aren't binary compatible with the 5.0 libs you had
08:19.03SexyKenNow I need to find out how to remove off and install:  /var/cache/apt/archives/libmysqlclient15_5.0.16-1_i386.deb
08:19.07fourcheezeSexyKen: best idea is to rebuild you asterisk using the new libs
08:19.10tzafrirWhat Debian exactly?
08:19.11SexyKendpkg complains if I do dpkg -i /var/cache/apt/archives/libmysqlclient15_5.0.16-1_i386.deb
08:19.26SexyKenNot 100% sure
08:19.48tzafrirAnd where did you get mysql5 from?
08:20.14fourcheezeapt-get install libmysqlclient15
08:20.22SexyKenWont let me
08:20.25SexyKenI dont have mysql5 at all
08:20.27SexyKenJust the libs I guess
08:20.54SexyKenE: Package libmysqlclient15 has no installation candidate
08:21.11tzafrirSexyKen, again, which Debian exacly? Sarge (Stable)? Etch (Testing)? Sid (Unstable)?
08:21.19SexyKenStable I believe
08:22.05tzafrirSo why do you bother with mysql5? Stable has 4.1 and 4.0
08:22.40SexyKenI didn't bother with 5
08:22.42SexyKenGah
08:22.44SexyKenYou're not hearing me?
08:23.13fourcheezeSexyKen: the point is you had the 5.0 libs installed and someone must have built asterisk
08:23.30fourcheezenow there are different libs the rug has been moved from under its feet
08:23.36tzafrirIs it a self-built module, then?
08:27.03*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)
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08:39.27creadurexanyone here used the ip10s and sip firmware?
08:44.59*** join/#asterisk backblue (n=igor@82.102.1.42)
08:51.58*** part/#asterisk shanky (i=jramirez@217.11.114.145)
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08:58.15*** part/#asterisk ernie_r (n=ernie@162.sub-70-203-21.myvzw.com)
08:58.46coppice"Wildlife Defies Chernobyl Radiation" - they probably aren't being defiant. they probably just can't read the keep out signs. :-)
09:00.10zoahehe
09:00.38zoamaybe they will start writing softmodems after a while :p
09:01.18*** join/#asterisk ionix (n=ionix@softbank221083058014.bbtec.net)
09:01.19zoai wonder how long it will take for the birds to spread the contamination
09:01.22ionixHello
09:01.36zoai mean if they keep eating and dropping some stuff on other places
09:01.43zoain the end it should be nicely spread, no
09:01.45zoa:)
09:01.57*** join/#asterisk ptblank (n=MURDER1@68.233.145.253)
09:01.58zoawe should put nuclear waste in the whitehouse
09:02.02zoato keep the terrorists out
09:02.04ionixI am trying to make outgoing calls to about 250 phone numbers (about 20 at the same time) for school closure.
09:02.27ionixI am trying to make a PHP script take up when the party answers and the PHP would control asterisk.
09:02.36ionixthe problem is that I am not sure how to debug it
09:02.52ionixhow can I see in real time the execution of the php script?
09:03.05*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
09:03.09*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
09:03.13Bert-hello there
09:03.19Bert-i've a question please
09:03.22ionixi.e See the print ""; when asterisk is calling it. Is it a option on the console or I have to output to a file and tail -f the file?
09:04.06Bert-I want to terminate some call. But i'll receive both G723 and G729 calls. How can I deal with ?
09:04.27Bert-because i know G723 is only free for passthrought use
09:04.38coppicezoa: they already put crap in the whitehouse, so the basic scheme for waste handling is already there
09:04.53Bert-and I wonder if i can setup two codecs
09:05.40Bert-then Asterisk can deals with one or other
09:05.52*** join/#asterisk [Airwolf] (n=airwolf@195.80.226.242)
09:10.20zoacoppice, some mutations might be good there :)
09:11.31backbluewhat do i have to configure in misdn to connect a phone and have tone?
09:11.33*** join/#asterisk ckruetze (n=ckruetze@62.214.75.2)
09:11.42backbluei can make calls, but i dont have tone, does anyone knows?
09:19.03xbit`i have exten => _1.,1,Dial(SIP/${EXTEN},20,t) exten=> _1.,2,Hangup in my dialplan, but the phone beeps after someone hangs up, and the phone waiting for the user to manually hang up. why is that?
09:21.10*** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron)
09:21.18*** join/#asterisk potsboy (n=chrisg@196.34.241.242)
09:22.01potsboyhi all, can anybody recommend a p4 motherboard for use with two tdm400's
09:23.30ionixpotsboy: Hmm, how the f this has something to do with asterisk
09:23.54ionixPlease kepp the requests asterisk oriented. Thx
09:24.32*** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-41-54.w86-213.abo.wanadoo.fr)
09:27.13potsboya motherboard for a asterisk implementation
09:27.40potsboytdm400 cards.. have had a "TDM MASTER ABORT" error on improperly IRQ asignments
09:27.53potsboywhen loading the wctdm module
09:27.58hackeronpotsboy: we use a nforce4 amd motherboard and get blipping on one of the tdm400 cards probably because of a shared irq
09:28.28potsboyany nforce you can recommend running on intel
09:28.53hackeronany reason why intel over amd?
09:29.12potsboyalready got the cpu
09:30.22hackeronah, personally had terrible experience with trying to pick a motherboard for an intel cpu. We bought a motherboard that was certified for asterisk with 2 sata drives - as it turned out the sata controllers werent supported by linux, ARGH!
09:30.37Bert-well nobody can/want answer to me ?
09:30.58hackeronpotsboy: and it was a 1U, it wasnt like we could just not use the sata controllers...
09:31.04hackeronpotsboy: the drivers were hotplug
09:31.05coppicewhich SATA controllers are those?
09:31.27hackerongod knows, we sent it back and got an nforce4 motherboard with the amd64 3500+ :)
09:31.45creadurexdell <3
09:31.50potsboyits a 2u
09:32.26hackeronat the time I googled for the motherboard and found that there is a binary driver available for some ancient 2.4 kernel that is no longer supported, yet this was advertised as "designed for asterisk", shish
09:32.47coppicei wonder how PCI and PCI-E compare for the kind of issues afflicting these telephony cards?
09:34.53coppicenumerous boxes with no T.38 support are advertised as having it. you expect honest advertising for VoIP stuff? ;-)
09:35.10vgstersupermicro p8sct is a pain mobo
09:36.59coppiceall mobos are a pain. when they seem OK its just because you aren't using the bits that cause pain
09:38.04vgsterhad to run the te110 card inthe pci-e slot
09:39.16coppiceno you didn't. it wouldn't fit
09:40.14vgstersorry pci-x slot
09:40.32vgsterget confused with all these versions
09:40.59coppicepci-x will automatically back off to a lower speed for these cards
09:41.43vgstergood job too cos if i run the card in a normal pci slot i get hdlc errors
09:41.53vgsterbut not in the pci-x slot
09:44.00*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
09:44.37coppiceinteresting. is that because something else high bandwidth shares the other PCI bus? people always talk about PCI latency and shared irqs. people don't talk much about that is sharing the bus
09:45.06vgsteryes the pci-x irq was shared with eth0
09:45.41vgsterif it ran on the regular pci bus it had its own irq but saw lots of hdlc errors, didnt seem to make any sense
09:46.01coppicenothing else on the regular PCI bus?
09:46.05fourcheezewhat tricks do people use to prevent race conditions in dialplans?
09:46.09fourcheezecounters?
09:46.13vgsterif i disable eth0 so the card is in the pci-x slot and has its own irq then its fine
09:46.28vgsteri had a 3ware raid card
09:46.42vgsterwith a 64bit interface that i had to run in a 32bit slot
09:49.06fourcheezeI want to avoid things like this:
09:49.06fourcheezeexten => 1,1,Dial(Local/2,30)
09:49.06fourcheezeexten => 2,1,Dial(Local/1,30)
09:49.06fourcheezegoing on forever
09:50.02coppicedoes prevent race conditions == equal opportunity dialplan? :-)
09:50.08fourcheezeI figure I could use a macro instead of Dial() which checks for a counter
09:50.31fourcheezecoppice you might need to describe what you mean by "equal opportunity"
09:51.33fourcheezeI want to stop the situation where a call can be in an endless loop tying up resources
09:53.16fourcheezethere most be some techniques people use
10:01.04*** join/#asterisk apardo (n=apardo@241.Red-80-34-202.staticIP.rima-tde.net)
10:02.03*** join/#asterisk AsteriskAlbania (n=info@217.24.244.130)
10:02.23AsteriskAlbaniaHI to everyone
10:02.40creadurexerrm.. why arent noop() outputting anything to my CLI? verbosity is like 10000
10:02.49AsteriskAlbaniaany one experienced with asterisk and radius authentication ?
10:03.11*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
10:03.12AsteriskAlbaniais there any opeensource solution ?
10:03.41*** join/#asterisk suma (n=suma@222.165.112.215)
10:03.44AsteriskAlbaniaI have seen somee commercial solutions, but price is too high
10:04.31sumai have problem in configuring phpmyadmin for sql server
10:04.44sumacan anyone please help me !
10:04.45AsteriskAlbaniasuma: tell me the problem
10:04.58AsteriskAlbaniasuma: I have installed it several times
10:05.04suma#1045 - Access denied for user: 'root@localhost' (Using password: NO)
10:05.17sumai hace changed config.inc.php to the server which i require
10:05.19AsteriskAlbaniahave you set a pass for your mysql server,
10:05.35AsteriskAlbaniaare you able to connect with mysql -p from console ?
10:05.35sumabut still it gets root@localhost
10:05.40sumayes i can
10:05.46sumaI can connect with c program
10:05.55sumaand console too
10:06.06AsteriskAlbaniaso you need to create config.inc.php at the root directory of phpmyadmin
10:06.30AsteriskAlbaniaare u using the latest verion of phpmyadmin ?
10:06.32sumai could access with php script that access remote mysql
10:06.39sumayes i'm using the latest version
10:07.06suma2.8.0.3
10:07.08AsteriskAlbaniado you have a real ip address at your machine that I can test from here ?
10:07.15AsteriskAlbaniahttp://ipaddres
10:07.31sumaphpmyadmin ip ?
10:07.38AsteriskAlbaniamachine ip
10:07.45sumahttp://www.kasangadu.com/115test/phpmyadmin/
10:07.50sumaor mysql machine ?
10:08.01AsteriskAlbaniaphpmyadmin
10:08.08sumathat is the one for phpmyadmin
10:08.12*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
10:08.15jhiver~seen anthm
10:08.24jbotanthm <n=anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 15h 54m 19s ago, saying: 'so it's a thin balance'.
10:08.31AsteriskAlbaniasuma: Wrong permissions on configuration file, should not be world writable!
10:08.44AsteriskAlbaniayou should change to 755 config.inc.php
10:09.19sumai did changed that now
10:09.59AsteriskAlbaniasuma: The configuration file now needs a secret passphrase (blowfish_secret).
10:10.45sumai will check it now
10:13.02AsteriskAlbaniasuma: Write some random text into the "secret" field.
10:14.30sumadone
10:15.43sumastill not working
10:16.36AsteriskAlbaniacan you check the auth mode is it cookie or http ?
10:16.47sumacookie
10:17.17AsteriskAlbaniatry to use http,
10:18.03sumachanged to http
10:18.08AsteriskAlbaniaok enter now
10:18.12AsteriskAlbaniait seems to work
10:18.20sumareally ?
10:18.45AsteriskAlbaniaI am asked for username and password
10:18.54AsteriskAlbaniatry it your self
10:19.13sumalet me check with other browsers
10:19.31AsteriskAlbaniatzanger: Hi
10:20.13AsteriskAlbaniatzanger: Do you know if there is a option for asterisk to authenticate and bill the calls to a radius server
10:20.42sumait is not working the given username and password
10:21.12sumausername and password is correct
10:25.25RGi_AsteriskAlbania : CDR records ?
10:27.08*** join/#asterisk ronn (n=zakforev@84-45-132-117.no-dns-yet.enta.net)
10:27.33*** join/#asterisk [Airwolf] (n=airwolf@195.80.226.242)
10:27.43AsteriskAlbaniaRGi_: I just need an interface to communicate with radius
10:27.51AsteriskAlbaniaI have a radius server for VOIP
10:27.59AsteriskAlbaniaRBS ALEPO VOIP
10:28.38key2!seen kram
10:29.50*** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com)
10:49.36*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
10:57.41*** join/#asterisk BearMan (i=karsten@freenode/staff/sourcemage.wizard.BearPerson)
11:03.44*** join/#asterisk shiznatix (n=shiznati@213-35-241-48-dsl.end.estpak.ee)
11:04.12shiznatixcan anyone tell me how to keep a certain terminal from having more than a certain amount of simultanious connections?
11:04.27*** join/#asterisk RoyK (n=roy@host-81-191-145-46.bluecom.no)
11:05.13jhiverhi everybody
11:05.37jhiveris somebody using Asterisk with g.729 + 40 ms packetization? I need to do a test with these values
11:05.58jhiverI have used anthm's patch because I have one way audio with a 40 ms gateway
11:06.18jhiverbut even after the patch I still have one way audio :-/
11:06.50tzangerI haven't ever played with the packetization
11:07.27jhiverI applied anthm's patch correctly, set "framems" in the sip.conf to 40, and still nuthin' :-/
11:07.57jhiverI have a SER -> Asterisk -> Audiocodes PSTN GW setup
11:08.05jhiverthe audiocodes is set to 40 ms
11:08.31jhiverAnd I get somebody to call me on my PSTN phone through SER
11:08.41jhiverI can hear the person but he can't hear me
11:08.46*** join/#asterisk psirac (n=psirac@217.175.165.34)
11:09.13psirachello all
11:09.19jhiverhi
11:10.26*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
11:10.55*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
11:12.07psiraci've a digium TDM02B card, every day i must restart asterisk and reload zaptel and wctdm modules because i've a big noise in my SIP Phone
11:12.45key2psirac: so put the reboot in ur crontab
11:14.12psiraci do this, but this problem come many times in a day
11:14.25psiracrandomly
11:14.44psiraci don't know what is it
11:16.07psiraci use the same hardware, with the same configuration in another place and i've no pb.
11:16.26mutugh
11:16.33muttelnet is having problems once again
11:16.40RGi_morn mut
11:16.43mutit's sad when a guy has had phone service for a month and a half
11:16.56mutand he calls in and says "THE LAST TIME THIS HAPPENED..."
11:17.11mutit was a switch on telnets end or something
11:17.36mutmor RGi_
11:17.37mutmorn
11:17.44tzangerkey2: no, you don't do that
11:18.11tzangerpsirac: call Digium support.  Be ready to give them root access to your system so they can diagnose this.
11:18.30tzangerpsirac: describe "a big noise in my SIP phone" ?
11:18.40tzangerpsirac: what version of asterisk is this, by the way?
11:19.12tzangerpsirac: have you swapped the TDM02B cards to see if the problem follows the card or stays with the system?
11:19.23RGi_anyone use FAX with asterisk here ?
11:19.48psiractzanger: 1.2.6 for asterisk, 1.2.5 for zaptel, yes i've swapped the cards
11:19.58tzangerpsirac: swapped the cards and the problem stays with the machine?
11:20.02tzangerand not the card?
11:20.32tzangerpsirac: e.g. Box A with Card A = noise.  Box B with Card B = no noise.  Box A with Card B = ?
11:20.39tzangerand Box B with Card A = ?
11:21.44tzangermut: you should know not to use telnet... change to the ssh provieder.  :-)
11:21.52psiracBOX A with card A = noise, Box A with card B = noise, Box B with card A = nonoise, Box B with card B= no noise
11:22.10muttzanger: if only it were the same context ;P
11:22.38tzangermut: I could see a real geek marketing opportunity here.  start a clec called ssh in the same areas as telnet and market it as "we're more secure!"  :-)
11:22.44shiznatixhow can i keep a terminal from putting like 10 people on hold at a single time? like i want to limit it to being able to have 3 connections at once, how can i do this?
11:22.53mutand reliable
11:22.55mutheh
11:22.58tzangershiznatix: what technology?
11:23.06tzangerbtw, cool nick.  :-)
11:24.59mutyay
11:25.13mut"i dunno the best way to spin this, maybe extraneous circumstance"
11:25.22mutwhat they just call me and tell me
11:25.59psiractzanger:  there is a little difference with BOX A and BOX B: BOX A have a DUO analog lines, and BOX B have TWO analog lines. With DUO you've two lines and two interfaces with one  phone number only. When you call this number if nobody reply on interface one, interface two ring
11:26.00tzangerwhere's the BOFH excuse calendar when you need it
11:26.10mut~bofh
11:26.13jbotrumour has it, bofh is Bastard Operator From Hell - see http://www.theregister.co.uk/odds/bofh/
11:26.18mut:P
11:26.27tzangerpsirac: ok but you still have a TDM02B in each box, right?
11:26.51psiractzanger: right :)
11:26.51tzangerjust Box A has two POTS lines that are equivalent.  If one is busy it rings down to the second number, and if that is busy the caller receives a busy signal
11:27.00shiznatixtzanger, just like a SIP or a IAX2 phone. just somthing in the dialplan that could be like 'you already have 3 connections, no more allowed' or somthing
11:27.14tzangerpsirac: ok, so the question remains:  Box A with Card A = noise.  Box B with Card B = no noise.  What does Box A wiht Card B and box B with Card A give you?
11:27.30mutwell i definatly need some coffee
11:27.37mutgotta run to the other office
11:27.48psiractzanger: same, Box A=noise and Box B no noise
11:28.56tzangershiznatix: simple:  each caller in their own context to make things simple, then a GotoIf(DB(UserACalls) > 2)?nomore), Set(DB(UserACalls)=$[DB(UserACalls)+1]), Dial(), Set(DB(UserACalls)=$[DB(UserACalls)-1])
11:29.06tzangerpsirac: ok so it's not the card.
11:29.44tzangerpsirac: next thing to try, just to eliminate this variable.  Physically move BoxA/CardA to BoxB's location and vice-versa.  Swap the HDDs so that Box A's HDD is in Box B with no other changes
11:30.07tzangerpsirac: we're trying to make sure that there is nothing screwy with this DUO line.  I really doubt it, but just to save us from chasing ghosts, we need to test that.
11:30.59tzangershiznatix: did you catch what I'm doing?  I'm checking a DB var before dialing and refusing to complete the call if the DB is saying they have more calls than you are allocating.
11:31.15shiznatixis there any way to do this without using a DB?
11:31.18tzangershiznatix: then, I increment the var, make the call, and decrement the var after the call.  You need to make sure that Dial() has the 'g' option in it
11:31.54tzangershiznatix: and to be safe, I'd add an 'h' extension which decrements the var *IF* it is not already zero, and sets it to zero if it's less than zero.  (basic housekeeping)
11:32.05tzangershiznatix: why?  This is dead-simple
11:32.07mutugh
11:32.08mutit's locked
11:32.16mutand the key isn't where it's supposed to be
11:32.26shiznatixtzanger, yes i know what you mean but we can't use a database for this sadly
11:32.42tzangermut: break a window, it's not your fucking problem if they can't be arsed to keep things where they belong
11:32.59tzangershiznatix: uh, this is the internal Asteirsk DB, not MySQL or PG or anything.  It's already installed and in use for other things.
11:33.11tzangeryou can use an external DB but it's really not necessary here
11:33.31muteh i'll just get a key for my keyring i guess
11:33.41tzangermut: if they complain, ask them if they're rather replace a window or have the system down until some fucknut comes in to unlock it?  :-)
11:33.44Drukentzanger: why not just use the outgoinglimit ?
11:33.54mutheh
11:34.00mutwell it's the techsupport/accounting office
11:34.04mutthey got the bunn coffee maker
11:34.06tzangerDruken: does that always work?
11:34.11*** join/#asterisk De_Mon (n=de_mon@fl-69-69-140-189.dyn.sprint-hsd.net)
11:34.16muti got some POS here that leaks and takes an hour to brew 4 cup pot
11:34.22Drukentzanger: does for me....
11:34.26mutcan't use that excuse
11:34.57tzangerDruken: actually it's call-limit in sip.conf and no such thing exists in iax.conf... this is technology independent
11:35.16coppicehum. a Point Of Sale coffee maker. a nice cup of coffee which queuing at the checkout. sounds a good idea :-)
11:35.28tzangercoppice: no ... you know how nasty that coffee'd be?
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11:35.48tzangerDruken was chewing his coffee yesterday, I imagine that stuff you'd have to lick like a lollipop
11:36.03mutcoppice yep, processes credit cards and even checks
11:36.15tzangerthese F1000G phones kind of suck.  better than than the WiSIP though
11:36.20muthaving an issue with the credit card company not wanting to process master card debit cards tho
11:36.25mutbeen working on that a month
11:36.35Drukenhmmm.... i wonder...
11:36.40mutand their tech hasn't called me for a week, and i've left 3+ voicemails everyday
11:36.44mutargh
11:36.51coppicesounds like a customer for a soft V.22bis in * :-)
11:37.00tzangerfind another vendor
11:37.10mutknow any?
11:37.19mutthat use verisign for a gateway
11:37.39mutthey already told me they're waive all my cancellation fees
11:37.44mutthey'de
11:37.45tzangercoppice: did you and redder86 get that v17 transmit working?  Last time I was looking at iaxmodem to talk to my shitty Xerox and Canon faxes you and/or he had said that v17 would work better after you'd done something to it
11:37.51mutthey'd
11:37.58Drukenmut, coppice, tzanger: can you guys all call my 866 number? i want to see if my iax will take 3 calls, i've never actually tested the incominglimit settings and stuff
11:38.07mutuh sure
11:38.09muttis it
11:38.26muti got 4 phones in here..
11:38.28Druken866-473-2121
11:38.33Drukensweet :)
11:38.41tzangerringy ringy
11:38.51tzangerABS something network?
11:38.51coppicetzanger: we got V.29 staggering along with the crap from the canon faxes. their modem is really badly broken. V.17 still needs completing
11:39.23mutthere ya go
11:39.24tzangercoppice: ahh okay
11:39.25muti got 4 calls
11:39.27Drukenhmm.... look at that
11:39.28tzangerI got 1
11:39.30mutit's like delayed stereo
11:39.31mutheh
11:39.57Druken:)
11:39.59mutthat enough for ye?
11:40.02Drukenyup
11:40.04mutlong msg
11:40.07mutor did it repeat?
11:40.13tzangerwhoever recorded these greetings sounds like they'd do well selling insurance... so nice and calm and not overly cheerful but still happy
11:40.17Drukentzanger is right, iax has no limiting
11:40.18psiractzanger: same problem but problem come randomly, sometimes at the end of a few hours, other times at the end of a few minutes
11:40.36Drukenwho's the 519?
11:40.38tzangerpsirac: ok, but at location A only?
11:40.40tzangerDruken: that's me
11:40.42tzangerI just hung up
11:40.42shiznatixtzanger, sorry for the delay but thanks for the info!
11:40.43Druken:)
11:40.48tzangershiznatix: no problem
11:40.54Drukenmut must have been the 989
11:40.58mutyep
11:41.03tzangerDruken: who's your 866 provider?
11:41.12Drukenwhy?
11:41.21tzangerDruken: because I need one
11:41.28Drukenahh :)
11:41.37mutahhh sweet
11:41.37psiractzanger: yes only with DUO interfaces
11:41.43mutthe boss had pepsi in his fridge
11:41.45mutthat'll hold me over
11:41.53Drukencokes better :)
11:42.05mutthere mt dew too
11:42.09tzangersivana runs voctel.net and has great 800 rates but his carrier will not give me a list of numbers that are available.  not his fault, but impossible to get 800#s for my clients since the ywant to see what is available to pick
11:42.18tzangerinstead of randomly plugging in numbers to see what's available
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11:42.50Drukenin otherwords, you want a block of numbers then
11:43.36tzangerDruken: no, I want the list of what's available at the point in time that I ask, so I can present that list and say "which number(s) are you interested in?"
11:43.36tzangernot "here's a web interface where you can punch in what you want and get disappointed 8000000000 times"
11:43.38Drukenwhats richard giving ya for pricing?
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11:43.56tzangerI mean I'd even go as far as to flood their server with requests to screen-scrape the results and build the table, but that's a last resort
11:44.13tzangerDruken: I'd have to check since I don't see the bills but I *THINK* $0.055
11:44.13*** part/#asterisk jtar (n=jtar@81-178-54-37.dsl.pipex.com)
11:44.37tzangerI'm just the tech guy, the bills go directly to the accounting department :-)
11:44.40Drukenwow, that's excelent
11:44.58tzangermakes it nice when I order $3500 in parts from Digikey or a $5000 piece of equipment... makes me feel less guilty :-)
11:45.16tzangerI forget if that $0.055 is richard or nufone or asterlink
11:45.23tzangermay be all or none, too. heh
11:45.29Drukenand to think, i can remember him comming in here asking questions about how to setup his pri card :)
11:45.37tzangerand the price was different for CAD-only vs CAD/US48 vs CAD/US48/US50
11:45.51tzangerDruken: we're all newbies at some point
11:46.06Drukentoo true
11:46.06shiznatixtzanger, in your sample code stuff you have 'UserACalls' but could I replace that with ${EXTEN} ? Sorry for the interogation here
11:46.07tzangerI turned that man into a database guru :-)  his billing/provisioning system is *SWEET*
11:46.32Drukeni seen the flowchart, and omg
11:46.37tzangershiznatix: no, because ${EXTEN} is the number they're calling.
11:47.42tzangershiznatix: ${ACCOUNTCODE} would be appropriate if you are setting accountcode=foo in iax.conf and sip.conf
11:48.09tzangershiznatix: ${CHANNELNAME} may work too, I'd Nop(CHANNELNAME is ${CHANNELNAME}) before to make sure you're getting the vlaues you're expecting
11:48.17koenviany one experienced in Dial() with G and D option?
11:48.28Drukentzanger: i don't really have a billing/provisioning system... :( i need one :)
11:48.53tzangerDruken: me too
11:49.00tzangerDruken: so who is your 800 provider?  is it richard?
11:49.05Drukennope
11:49.09Drukenthinktel
11:49.22tzangernever heard of 'em
11:49.35Drukenalberta
11:49.39tzangerunlimitel rocks, I use them for DIDs and termination for on-net
11:49.48tzangerI am *REALLY* happy with Unlimitel, in fact
11:50.05Drukeni use a bunch of carriers for termination
11:50.08tzanger$0.011 is good enough for me, I don't go for the absolute cheapest... same reason I use nufone/asterlink
11:50.25tzangerI use unlimitel where I can, then nufone/asterlink, then fall to PRI
11:50.57Drukeni have a few default routes, with failovers to some others
11:51.10Drukenif i have a local route to it, of course i'm gonna use it first :)
11:51.14tzangerha.  thinktel's contact us link is broken
11:51.21tzangerDruken: yep that's what I do... poor man's LCR
11:51.23Drukenare you shitting me?
11:51.32tzangerDruken: about what?
11:51.37Drukenthe contact us
11:51.38Drukenhehe
11:51.40tzangernope I'm not
11:52.00vgsteranyone use a sip/iax provider in the UK for cheap calls?
11:52.09Druken780-424-3303
11:52.24Drukenbut they aren't open till 8-9 alberta time
11:52.26tzangerif you want maybe we can do some peering for our respective local calling areas (I have Kitchener/Waterloo area)
11:52.32Drukenso 11 est
11:52.35tzangeryeah
11:53.10koenvican I use the G and D option together in the Dial application???
11:53.16Drukentzanger: i have all of ontario, montreal, edmonton, calgary, and a few others
11:53.28tzangerDruken: ??  for a local (TDM hopoff) calling area?!
11:53.40Drukenoh god no
11:53.42Drukenhehe
11:53.55Drukeni don't have a single tdm hopoff
11:53.59koenviI want to send DTMF when my call is answered and then go on in another context
11:54.09tzangerthat's actually one of my requirements for a termination provider... they must do the TDM hopoff, I won't go through a reseller just due to network hassles
11:54.45tzangerand it looks like unless nufone can find or renegotiate with their telco I will stop using them once my account's dried up
11:54.56Drukenwell, i'm on the local network with thinktel, on a ds3 that includes the 400 corridor
11:55.06tzangerwell yeah that's a little different
11:55.10tzangerunlimitel's like that too
11:55.17shiznatixtzanger, where can i find a list of the global variables that I can use for such things like a definition for ${EXTEN} and the like?
11:55.27tzangerasterisk source/doc/channelvariables.txt
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11:56.09Drukentzanger: i'm sorta like you, all my carriers do the PSTN work, instead of hand them off
11:56.21*** join/#asterisk saftsack (n=saftsack@p54A7E42E.dip.t-dialin.net)
11:56.32tzangerDruken: yep :-)  cheap is good, but if I have to endure complaints from the 30 people here I'll drop the carrier
11:56.45brif8I'm using  tftp-hpa for my TFTP Server.  The connection is "made/attempted" but the file is not sent/received?   TFTP Server --- WAP --- PC,  why ?
11:57.03mutheh
11:57.10mutfree service
11:57.14Drukenhehe because of the failovers in my dialplan, i've never had a call not go threw
11:57.32mutseems almost every install we do we give the person free service because it's never working correctly when they say it is
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11:58.11tzangerDruken: same here.  but that's not what I'm worried about.  I'm worried about shitty (but not down) networks... wwwwwwww h h h enca l l ls st t t t st  st st arttttt soo o o o ound ding l l l l liik k k k k e thi s s s s s s s s...
11:58.20shiznatixtzanger, would it make more sence to use ${CHANNEL} instead of somthing like ${CHANNELNAME}??
11:58.48Drukenthat took some tallent to read :)
11:59.03tzangershiznatix: not sure, you have to use Nop() to make sure that you're getting a value that is always the same for that SIP/IAX user.  You don't want IAX2/user-5 because that "-5" changes all the time.  you want IAX2/user or just user
11:59.15blitzragemorning all
11:59.17tzangerDruken: and it's mighty annoying to endure for a non-techie who just wants to use the fucking phone
11:59.47Drukentzanger: i got one of those...
11:59.55tzangerI've got 30.  :-)
12:00.01tzangerand one of them writes the cheques I depend on
12:00.11Drukenbut luckily they are generally very happy with the service, cause they save shitloads of money
12:00.17tzangeryup same here
12:00.25Drukeni just wish they wouldn't save 400 messages in their god damn voicemail
12:00.25tzangeronce bell stopped overcharging us for hte PRI
12:00.40tzanger(got an $18k cheque from them, and a separate $4000 one after I got it fixed, if you can believe it!)
12:01.03Drukenwhy can't i get some of those?
12:01.04Drukenhehe
12:01.10blitzrageDruken: you can put a limit eh? :)
12:01.36Drukenblitzrage: on ?
12:01.42blitzragevoicemail
12:01.53Drukenoh yeah?
12:02.10Drukeni know it's only 100, generally, but they figured out the voicemail has "folders"
12:02.11drrayshouldnt be hard to hack up an auto delete voicemail setup
12:02.45blitzragecheck voicemail.conf, I'm pretty sure you can -- and there is a script in the asterisk source directory to delete old emails automatically
12:02.51Drukeni don't know why anyone would do that in the first place...
12:04.03Drukeni can see keeping emails, but voicemail like that? my god... could you imagine looking for something? it'd take ya all day
12:05.14koenviI want to send DTMF when my call is answered and then go on in another context
12:06.15*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
12:06.46koenviusing Dial(,,D()) works fine, Dial(,,G()) too
12:06.55blitzragekoenvi: SendDTMF() ?
12:07.10koenvibut the combination Dial(,,G()D()) doesn't
12:07.32koenvihm ... would it be that simple
12:07.36blitzrageoh yah, you were talking about this the other day -- you mean after the other end picks up
12:07.49koenviindeed
12:08.15koenviboth options will do something after the other end pick up
12:08.49koenviI guess they are mutualy exclusive
12:09.23koenviI'll give the SendDTMF a try
12:10.03blitzrageyou could probably use SendDTMF and then use the M flag to run a macro once the call is picked up
12:10.14blitzrageI think it's M anyways
12:10.49koenvidoes the M flag send both legs to the macro?
12:10.59blitzragekoenvi: not sure
12:11.38*** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx)
12:11.54Mikeguys, digium g729 licenses work with ser also?
12:12.03blitzragedoubt it
12:12.09LostFrogser doesn't do media, does it?
12:12.16blitzragewell -- actually, I guess they do -- because SER doesn't do media :)
12:12.33blitzrageSIP signalling is separate from the media
12:12.57Mikeso how exactly does it work? ser + asterisk?
12:13.03blitzragemagic!
12:13.16blitzrageSER handles registration and signalling, then you send calls to Asterisk when you need to
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12:13.34Mikeyes but i see that ser can handle 30 call each second
12:13.36blitzrageAsterisk then becomes a feature server, and only goes to Asterisk when it needs to
12:13.44Mikeasterisk cant handle 30 calls even in 30 seconds
12:13.50Mikehow come you can use its power?
12:14.01blitzrageMike: but if you need to do transcoding, you have to send the call to Asterisk
12:14.19blitzrageMike: doesn't handle 30 calls in 30 seconds? pffffft -- if it didn't, it'd be pretty hard to run a VSP
12:14.34Mikefor example if i have 10 sipuras using g729 i can use only ser, and send them to asterisk just for pstn?
12:15.02blitzrageMike: yes -- as long as the other ends you're negotiating with don't need transcoding (i.e. everything speaks g729)
12:15.13*** join/#asterisk ic (n=ic@staff.rbi.speka.net)
12:15.15ichello
12:15.20Mikeasterisk? with g729? difficult to go 200+ calls at the same time
12:15.31icis there a way to have some information on the reason of an 488 error ?
12:15.33Mikeprocessor is almost at 10.00 average load
12:15.35*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
12:15.40blitzrageMike: depends on the processor
12:15.51Mikexeon 3.0 ghz
12:15.59blitzrageMike: sounds like you're transcoding then
12:16.12Mikeim going from g729 from sip to iax gsm
12:16.22blitzragewell there you go -- SER helps you not at all
12:17.08blitzrageyou're going from one compressed codec to another compressed codec, and trying to do 200+ sim calls, that's why your load avg is so high
12:17.14LostFrogI think I may have to go to SER to get hints working across IAX.
12:17.24Mikeok so, with ser i can take asterisk dialplan and still make ser handle the call?
12:18.35Mikeie use 200+ sipuras using g729 calling each other with ser but taking asterisk dialplan?
12:21.10blitzrageMike: SER is independent of Asterisk for routing -- so you have to create your own "dialplan" in SER to handle the calls going between each other
12:21.41blitzrageMike: you can send it to Asterisk, but then you don't really get any advantage at all
12:22.03blitzrageMike: just use canreinvite=yes for the phones so media goes to the phones directly and not through Asterisk (when there is no transcoding involved)
12:22.17Mikeok
12:22.38blitzrageI don't think you actually NEED SER at all unless you're doing a lot of registrations (thousands)
12:24.05Mikemaybe i could use it to load balance with alot of servers
12:26.59coppiceMike: do you really need to mass transcode between G.729 and GSM? the quality really sucks when you do that
12:28.05darkskiezif you use ser, can it allow you to have multiple devices register with the same details?
12:28.36darkskiezdo you have to move all of your authentication into ser then, or can it just proxy that to * ?
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12:33.02brif8is it possible that TFTP can fail due to NAT ?
12:33.02*** join/#asterisk RoyK (n=roy@217.8.134.129)
12:33.02shiznatixwhen comparing two variables in the extensions.conf should I use = or ==
12:33.03brif8= is fine
12:33.12Mikecoppice, gsm doesnt take that much procesor
12:33.37Mikecoppice, the box in the other end is smaller and cant handle something like 200+ g729
12:33.44coppiceI said nothing about processing power. I said if you transcode the quality sucks
12:34.17Mikecoppice, then i send it via iax then to a unicall channel
12:35.02Mikecoppice, one question on unicall, how can i make it not to reset channels evey once in a while?
12:35.17*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
12:35.22coppiceit doesn't reset channels
12:35.24Mikesometimes i have 30 full channels and it resets all of them droping the 30 channels
12:36.25coppiceno idea why. i haven't had complaints like that before, and a number of people push multiple E1s quite hard
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12:36.44coppicehow often does that happen?
12:36.57Mikecoppice, not just me, theres about 5 other people on a forum experimenting this
12:37.08*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
12:37.20Mikecoppice, sometimes every 2 hours sometimes ever 3 hours dunno
12:37.29coppicemaybe they should have reported it to me, instead of just whining amongst themselves
12:37.34Mikecoppice, happends it just drops all the calls on every channel
12:37.40blitzrageanyone ever get ANSI colour codes into the dialplan before? (NoOp())
12:38.00*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
12:38.01blitzrageI've seen someone do it before -- really wish I had made a note of how they did it....
12:38.09coppiceMike: do new calls then pick up OK, without you taking any action?
12:38.12Mikecoppice, its mexico variant, so maybe its just the variant, im using cpe as signalling
12:38.35Mikecoppice, yes they pick ok once the channels are drop
12:38.50shiznatixtzanger, can you check this dialplan and see if it would work? http://pastebin.com/673163
12:38.51coppicethe co/cpe selection is actually meaningless for MFC/R2. it is completely symmetric
12:39.28[TK]D-Fendershiznatix : Your GotoIf's are all wrong
12:39.33Mikecoppice, actually when i had co as a signalling i had 3 locks on the server every day
12:39.42[TK]D-Fendershiznatix : evaluations should be in $[]
12:39.44Mikecoppice, just change it to cpe and its been almost 5 days up
12:40.15[TK]D-Fendershiznatix : And your DB reads aren't right either
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12:40.35Mikecoppice, im using 0.0.3 from march 15 or so
12:40.50coppiceI can't remember anything in unicall which would be affected by the cpe/co selection for MFC/R2.  it should be completely ignored
12:41.14[TK]D-Fendershiznatix : Not to mention that I don't believe you will ever get a DB entry to match a channel....
12:41.36Mikecoppice, there is, im not the only one who says so
12:42.43coppice"TerminatingT38(tm)" how the bloody hell can you trademark that?
12:43.08Mikecoppice, i could send you a log when i get a drop on channels
12:43.19coppiceplease do
12:43.42Mikecoppice, loglevel 255 ok for you?
12:43.48coppiceyes
12:43.54Mikeok
12:44.28shiznatix[TK]D-Fender, sweeeeeeeeet. Can you give me some advice as I don't know what you mean really.
12:44.39shiznatix[TK]D-Fender, What evaulations are you talking about?
12:45.52[TK]D-Fendershiznatix : Well you really need to read up on FUNCTIONS on the wiki, as well as how to do GotoIf.  Second, what exactly are you trying to do there anyways?
12:47.03coppiceMike: I just searched through the source code. the co/cpe selection in unicall.conf is completely ignored when you are using MFC/R2
12:47.57shiznatix[TK]D-Fender, I thought I did read up on them but aparently not enough. I am trying to limit the amount of calls that a specific terminal can have at once so they can't put like 10 people on hold for like 20 minutes or somthing stupid
12:48.46Mikecoppice, worked for me :(
12:49.06Drukencome on 9:00!!
12:49.08Mikecoppice, can you tell me what linux distro, kernel, and unicall version do you have
12:49.11Drukenalmost there
12:49.23LostFrogDruken: Why?
12:49.27[TK]D-Fendershiznatix : Put "incominglimit" and "outgoinglimit" in their sip.conf entries....
12:49.33Drukenthen i can go have a shower
12:49.58coppiceMike: what I use may not be very relevant. people in south america test this stuff far more than me.
12:50.20coppiceMike: could the calls be dropped because of a small hiccup on the E1?
12:50.25LostFrogThank god, Druken. The EPA was calling. :)
12:50.28[TK]D-Fendershiznatix : Sometimes "h" just doesn't work right and you might get stuck.  Second, you are setting a DB entry that is unique for each CALL, not just each USER.
12:50.41Mikecoppice, what do you mean with a hiccup? and how do i prevent it?
12:50.45DrukenLostFrog: fak off :)
12:50.47*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
12:50.56coppiceMike: I'm not sure what glitch suppression there is in the zaptel driver for the CAS signalling bits
12:51.05LostFrogDruken: Gladly.. I spend most of my day doing that anyways.
12:51.24[TK]D-Fendershiznatix : CHANNEL would hold something like "SIP/5000-e63d" for a given call by SIP/5000 for example and wouldn't create a valid counter source.
12:51.25DrukenLostFrog: gotta wait till someone is in the office before i can leave my phones unmanned :)
12:51.27coppiceMike: by hiccup I mean a few bad frames of data, which might make the signalling clear the calls
12:51.28Mikecoppice, do you want to see my zap configuration?
12:51.56shiznatix[TK]D-Fender, The incoming outgoing stuff looks promising but what about iax terminals?
12:52.08[TK]D-Fendershiznatix : this is something that you would also most likely want to do in an AGI somehow first as verifying these sort of things is hard in the dialplan.
12:52.24[TK]D-Fendershiznatix : "iax terminals"?
12:52.27coppiceif it works most of the time it isn't likely to be the config file. try to send me a log of the calls dropping. I'll study that, and we can proceed from there
12:52.50Mikecoppice, maybe i could find something on the messages log let me check
12:53.05shiznatix[TK]D-Fender, "iax terminals" like in the iax.conf file...
12:53.27shiznatix[TK]D-Fender, do those incoming and outgoing things work for iax?
12:54.12Drukenno
12:54.13[TK]D-Fendershiznatix : Did you LOOK? (rhetorical question)
12:55.45[TK]D-Fendershiznatix : And a though : use the "g" Dial option instead of the "h" exten.  healthier that way.
12:56.20tzangershiznatix: sorry I got taken away fromthe system here for other work
12:56.43Ariel_morning folks
12:56.50Ariel_hope everyone is doing fine today.
12:56.58Drukentzanger: masterbation is not "other work" :)
12:57.14blitzragelol
12:57.21tzangerDruken: hahahaha
12:57.24Mikecoppice, whats this? Apr  4 20:41:42 WARNING[1174] chan_unicall.c: Unicall/27 Whoa....  I'm owned but found (45) in read [0x820c438, 0x820c438]...
12:57.46coppiceMike: sounds bad. do you have many of those?
12:58.19tzangerok what the hell driver do I need to get aaccess to /dev/sda in a XenU/
12:58.24Mikecoppice, some, and also some alarms
12:58.32Mikecoppice, should i send you the alarm logs?
12:58.50coppicesend the whole thing, and I will take a look
12:59.04Mikethe log is 30megas big
12:59.05Drukenwhat is unicall ?
12:59.25coppiceMike: this is why we have broadband these days :-)
12:59.29Mikecoppice, its to big, ill send just the intresting stuff that i can find now to your email, just sent you the alarms
12:59.41coppiceDruken: one less than Duocall, I guess
12:59.44Mikecoppice, how do i send you 30m logs?
12:59.49mutman
12:59.53Mikecoppice, i can give you access to the server?
12:59.54tzangerMike: carefully
13:00.04muthate when ya stretch and your hip/tail bone pops
13:00.10mutfeels so good tho
13:00.16Drukeni'm sure your irc client has dcc?
13:00.23coppiceMike can you put it somewhere I can FTP or HTTP from?
13:00.26tzangermut: I have one worse for you -- the bone in your chest that all your ribs connect to... when that pops it feels... WEIRD
13:00.33Mikecoppice, sure let me .gz it
13:00.37Drukenmut: i hate cracking my knees...
13:00.38mutnever had that
13:00.40mutdidn't think it could
13:00.43*** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no)
13:00.48tzangeryes my knees crack
13:01.06Drukentzanger: that one hurts like a bitch when that one goes (the chest)
13:01.12mutman my gf's do every time she moves
13:01.14mutit's crazy
13:01.16Drukeni've only done it once.. and oh my god...
13:01.17*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
13:01.17LostFrogtzanger: your sternum cracks?
13:01.21tzangerLostFrog: yep
13:01.22mutshe's going to have some problems down the road i bet
13:01.53Mikegzip make it on 2.2 m
13:01.57Mike:)
13:01.58LostFrogI didn't think it was possible.
13:02.04LostFrogMike: bzip2?
13:02.15*** part/#asterisk ic (n=ic@staff.rbi.speka.net)
13:02.16[TK]D-FenderLostFrog : With the right blade, all sorts of things are possible :)
13:02.33MikeLostFrog, gzip
13:02.40koenvi<PROTECTED>
13:02.58tzangerLostFrog: very possible.  chiropractor said that it can happen, as the rib/sternum connections are "knuckles" jus tlike fingers or other joints
13:03.08tzangerit's not NORMAL and if it continues it can be a problem but once in a while is not bad
13:03.14tzangerit used to be once or twice a day, it's almost never now
13:03.31koenviwhy do my calls move from one channel to the other on pri
13:03.47LostFrogkoenvi: they get restless?
13:04.14koenvifound some articles on this, but they talk about old bugs
13:04.43koenviand I'm running 1.2.6
13:06.09Mikecoppice, i think the drops are after i get the alarms "Alarm No Alarm raised, No Alarm cleared"
13:06.10koenvino pri specialists?
13:06.12blitzragemy knees crack too...
13:06.51LostFrogMy lips crack.
13:07.15blitzragesay no to crack!
13:07.21LostFrogDidn't even think about that, [TK]D-Fender.
13:07.26[TK]D-FenderI would never do a drug named after a part of my ass!
13:07.28LostFrogCrack kills.
13:07.35tzangerkoenvi: "move from one channel to the other" ?
13:08.01koenvi-- Moving call from channel 30 to channel 1
13:08.03[TK]D-FenderAhrimanes : Killing spree?
13:08.13Ahrimanes[TK]D-Fender: big ass crack pipe?
13:08.37muthuh what? crack?! wherE?!!?@
13:10.02*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:10.57*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
13:12.27*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
13:12.43coppice[TK]D-Fender: the chinese have a drug named after part of your ass - donkey penis broth :-)
13:13.15Mikecoppice, did you get the log?
13:13.54coppiceyep
13:14.03[TK]D-Fendercoppice : Watch your language... bkw_ lurks!
13:14.03Mikesearch for the alarms
13:14.13*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
13:14.17mutOMFG
13:14.20muti'm going to kill telnet
13:14.36blitzragewho the eff still uses telnet?
13:14.36mutthey had the problem fixed earlier
13:14.40mutnow it's broken again
13:14.42*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
13:15.01coppice[TK]D-Fender: he's strange. he kepts talking about moose penis, but don't seem to know what it tastes like
13:15.20mutblitzrage: you don't wanna know..
13:15.21LostFrogcoppice: better moose penis than mouse penis.
13:15.24coppicetelnet is extremy useful
13:15.25mutbest to stay away
13:15.37mutcoppice: sure if all you need is a t1 loop
13:15.58blitzrage:)
13:16.12*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
13:16.34mutit's sad when my VOIP over wireless is more reliable than their POTS
13:17.02*** join/#asterisk pengyong (n=lala@61.177.78.186)
13:17.09puzzledafternoon all
13:17.13muthow do
13:17.19austinnichols101morning
13:17.19mutoo man, thats some good coffee
13:17.30mutlike 9 scoops of sugar
13:17.46Ahrimanescaffeine AND sugar chock in no time?
13:18.41jsharpSign me up for that.
13:19.31blitzragehrmmm... any suggestions for a good way to send an email from the dialplan?
13:19.42*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
13:19.43Ahrimanesjsharp: yes please
13:19.46*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
13:19.57Ahrimanesmm almost luke warm coffe
13:19.58Ahrimanese
13:20.02blitzragebasically I want to send a short email from the DP if something BAD happens to an administrator
13:20.24Ahrimanesblitzrage: write app_email ? ;)
13:21.15blitzrageAhrimanes: I'd love to be able to have that ability :)
13:21.39blitzragemaybe I'll have to use a small AGI....
13:21.51blitzrageprobably the easiest
13:21.56*** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-47-119.w86-213.abo.wanadoo.fr)
13:22.16mutoo yea
13:22.21mutand sloppy joes for lunch
13:22.27muti might eat them early
13:22.34blitzragemut: are you on crack?
13:22.45muti'm high on life
13:22.49mut:P~
13:23.12Ahrimanesblitzrage: hm, should i use app_email as a learning project for my asterisk dev experience?
13:23.29blitzrageAhrimanes: I think it'd be very cool!
13:23.35Mikecoppice, found anything intresting?
13:23.47blitzrageAhrimanes: I'd test for you too if you want :)
13:23.50Ahrimanesblitzrage: lemme see if i can get it done over the weekend :)
13:24.03blitzrageAhrimanes: cool -- I'm on IRC all the time, so just come find me
13:24.14Ahrimanesblitzrage: :)
13:24.51coppiceMike: lots of alarms are cleared, but they never seem to be raised
13:25.04*** join/#asterisk Skarmeth (n=Skarmeth@201009048140.user.veloxzone.com.br)
13:25.13Vagabondcan I dial an Agent/XXXX? I can only get things to work if I use SIP/XXXX
13:25.29Mikecoppice, did you find those drops?
13:26.20coppicethere are a number of reloads of chan_unicall. why is that?
13:26.25brif8what would cause a 7920 not to pick up the OS7920.TXT file it connects to TFTP and gets the other files but not OS7920.TXT ?
13:26.42Mikecoppice, i never do "reload"
13:26.50Mikecoppice, if i want to do some change i restart asterisk
13:27.01coppicesomething is doing a reload
13:27.12Mikecoppice, maybe thats dropping my calls?
13:27.20coppiceprobably
13:27.27*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
13:27.33Ariel_quick question about static agents in queues.  Can someone explain what this last ,0 means in this line? member=Local/5124@from-internal,0
13:27.48Mikecoppice, maybe you shouldnt drop calls on a reload
13:28.22[TK]D-FenderAriel_ : Wow... chan_local for static agents... FREAKY
13:28.28Ariel_Vagabond, use the local command
13:28.48Ariel_[TK]D-Fender, works fine. but I am trying to see what the ,0 is for at the end.
13:28.54VagabondAriel_: can you show me an example?
13:29.12Ariel_Vagabond, see what I posted about the local
13:29.20blitzrageis there a dialplan function or application I'm not seeing that allows me to control which codecs to negotiate with during a SIP call setup?
13:29.53Ariel_blitzrage, there is a set you can do before you dial
13:30.10blitzrageAriel_: like a built in variable? I'm going to go check README.variables :)
13:30.49*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:30.54Ariel_blitzrage, since your an expert any Idea about the static agent: member=Local/5124@from-internal,0  <= trying to figure out the ,0
13:31.49VagabondAriel_: copuld it be the penalty?
13:31.54Kattyi think it's the queen's birthday today
13:31.55blitzrageI bet that's what it is
13:32.04blitzrageKatty: it is!!  ALL HAIL THE QUEEN!
13:32.04Kattyand i /think/ she's 80 today!
13:32.12blitzrageKatty: correct!! you win a cookie
13:32.16Kattyblitzrage: think she'll get cake?
13:32.19Kattyi want cake.
13:32.22blitzrageKatty: me too
13:32.29Ariel_Vagabond, maybe but could not find a good explaintion in the application nor on the wiki.
13:32.30Kattylet's have cake and celebrate!
13:32.32blitzrageKatty: I think I might go make brownies....
13:32.35blitzrageKatty: ok!
13:32.43coppice*the* queen? surely you mean *a* queen
13:32.56sivanahaha
13:32.56drfoomod2does anyone have a total access 750?
13:33.05blitzrageAriel_: ;member => Agent/:1,1           ; Any agent in group 1, wait for first
13:33.06blitzrage<PROTECTED>
13:33.06nahirean80 years old and still hot..  whew.
13:33.08fourcheezeI heard a rumour that there were other queens but I didn't believe it
13:33.19blitzrageAriel_: from agents.conf.sample :)
13:34.24Ariel_blitzrage, yes but normally we would use the /n option for local that is what is confusing me...
13:34.59blitzrageAriel_: you'd just need to add the /n to the end, the ,0 is still in regards to penalty
13:35.17blitzrageAriel_: although explicitly saying penalty zero seems to be unnecessary
13:35.24VagabondAriel_: maybe I'm just dumb, but how do I use this magical 'local' command? what'd I'd like to have is Dial(Agent/1000) work...
13:35.25coppicethere are still quite a few kings and queens around the world
13:35.42Ariel_blitzrage, yes I agree just trying to fix a problem with some queues... thanks.
13:36.11blitzragecoolio
13:36.22nahireananyone here familiar with the MYSQL addon for asterisk?  I am having a bit of difficult using MYSQL(Connect .. it's a simple DB that's setup, but it just "Fails"
13:36.43nahireandifficulty*
13:37.05nahireandoes it require a username and password?  i know it's in the syntax, but I am wonderig if it's possible to bypass it
13:37.13Ariel_Vagabond, Dial(Local/5124@from-internal,0/n)  would be what I am now testing
13:37.14blitzrageAriel_: ${SIP_CODEC}            Set the SIP codec for a call   <---- thanks for the tip!
13:37.23blitzrageAriel_: no -- that is wrong
13:37.26nahireanwondering* .. sheesh I need some mtn dew
13:37.36Ariel_blitzrage, correct just testing a few things
13:38.00blitzrageAriel_: oh -- and you're doing this in a Dial()?  then ,0 is actualyl in the place of the Dial flags
13:38.24blitzrageAriel_: Dial(Local/5124@from-internal/n)  <-- don't use ,0 at all
13:38.28fourcheezenahirean: only too familar
13:38.39Ariel_blitzrage, ok great thanks
13:38.54Ariel_just trying a few different things to see which one works best.
13:39.15blitzrageheh :)  yah, Dial() still needs the same format with Local channels as regular chanels
13:39.24fourcheezenahirean: it needs a password if you told it to need one
13:39.40nahireanfourcheeze, I can omit that in the syntax?
13:40.09fourcheezenahirean: I'm not sure what you're trying to do
13:40.28*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)
13:40.58fourcheezenahirean: I tend to do my mysql stuff through odbc
13:41.14blitzragesame
13:41.19fourcheezenahirean: where are you doing yours from?
13:41.21blitzragealthough I use postgresql :)
13:41.26*** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it)
13:41.32nahireanfourcheeze, I'm simply trying to connect to the database.. :) I am wondering if I can omit the username and password of the Mysql(Connect statement.. but everything I've read suggested that it's required
13:41.33Micettohi :)
13:42.00nahireanfourcheeze, I am trying to access a database from extensions.conf and read some things to a few variables ;) no biggie if I can actually connect ;)
13:42.10Vagabondnahirean: if it wants authentication, give it authentication
13:42.14Micettosomeone have the solution to corrupt 'clid' column in cdr rows with mysql ?
13:42.21*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.244.Dial1.SanJose1.Level3.net)
13:42.48MicettoI have this problem and I don't know to resolve it
13:43.05Micetto:(
13:43.50fourcheezenahirean: is that Mysql() in your dialplan?
13:44.09nahireanfourcheeze, Aye.. specifically: exten => 1,1,MYSQL(Connect connid localhost dbuser dbpass ml_test)
13:44.24nahireanset it up with that l/p but it's "Failing" for some reason
13:44.30fourcheezenahirean: I've never tried to do that
13:44.43Mikesomeone knows how can i return a 34 cause when my channels are all busy or inuse?
13:45.53*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
13:47.06koenvicapiinfo returns 'no such device'
13:47.21koenvicapi mod is loaded
13:48.21*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.244.Dial1.SanJose1.Level3.net)
13:53.34puzzledkoenvi: that can happen when there are no capi devices in /dev
13:54.33*** join/#asterisk MattH (n=MattH@63.174.244.195)
13:54.57MattHHi.... how can I debug MessageWaiting notifications?  Since upgrading to asterisk 1.2.6 my phones MWI no longer are working
13:55.25puzzledMattH: why would you want to upgrade to 1.2.6 if 1.2.7.1 is available?
13:55.45MattHwell this was before 1.2.7.1 was out
13:55.56puzzledexcellent reason :)
13:56.02MattHlol
13:56.18MattHand my understanding is 1.2.7.x doesn't fix any MWI bugs...
13:56.23MattHthough it does fix some memory leaks
13:56.39puzzledyup
13:57.02MattHso hehe.. back to the question :P
13:57.04Vagabondit still coredumps when I shut this machine down too :)
13:57.09MattHDOH!
13:57.31koenviexit
13:57.50Vagabond+
13:57.54Vagabondoops
13:59.56*** join/#asterisk techie (n=gus@antibala.net)
14:00.00*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:00.00*** mode/#asterisk [+o anthm] by ChanServ
14:00.12Kattymorning, anthm (=
14:00.17MattHI don't know if it would show in the full debug log... but I don't see asterisk tossing out a MWI notify
14:00.24brif8what would cause a 7920 not to pick up the OS7920.TXT file it connects to TFTP and gets the other files but not OS7920.TXT ? Or is there a way to force the phone to seek new firmware ?
14:01.58MattHINTERESTING!
14:01.59*** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com)
14:02.07MattHI had to put 'mailbox=extension' to get it to work in the sip config
14:02.45coppiceI wonder if * would run on this? http://www.linuxdevices.com/news/NS3880195342.html :-)
14:03.16*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
14:03.48LostFroglol.. It says "pecker."
14:04.45cpmheh, he said, ,
14:05.40anthmhi
14:06.21*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
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14:08.39Flautogood morning
14:09.39ManxPowerI still don't understand why do many people have problems with MWI.
14:09.53ManxPowerThe only time I have problems with MWI is when I typoed the mailbox= line
14:10.18coppiceits the 90V that gets them :-)
14:10.30b00mer_is chan_sccp ready for a production environment?
14:10.36*** join/#asterisk viLeR (i=1000@66.128.47.232)
14:10.51coppicei wonder how much of that 90V MWI stuff is in use
14:11.17jsharpSprint local used it in Florida.
14:11.38jsharpI got my bell rung, so to speak, a few times.
14:11.56Flautowhat do i need for voice reminder and wakeup calls? i mean to set up a time
14:11.59b00mer_i like the cisco 7920 and 7936 but it looks like only sccp software is out there...
14:12.11jsharpYeah, there's no SIP for the 7920.
14:12.44b00mer_how is the chan_sccp?  good?  I would be using it to supplement our sip production phone system
14:13.08Ahrimanesblitzrage: ?
14:13.12*** join/#asterisk saftsack (n=saftsack@p54A7E42E.dip.t-dialin.net)
14:13.27*** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk)
14:14.04ManxPowerI thought 90V was for VWMI 8-)
14:14.54brif8exit
14:14.56coppicewhat's a V between friends
14:15.15Greek-Boyhow do I get asterisk to phone skype users?
14:16.09b00mer_shifting gears... if I wanted to crete a method to allow someone to forward a number to one number and back to another number, by entering something into the keypad when the got to an office... what is the best way to do that.  the DB stuff?
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14:21.05blitzrageAhrimanes: yes?
14:21.13blitzrageGreek-Boy: you don't
14:21.22*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-54.claranet.co.uk)
14:21.27blitzrageGreek-Boy: skype is proprietary
14:21.38*** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)
14:21.52blitzrageb00mer_: yah -- just save data into the AstDB
14:22.03blitzrageb00mer_: might not be *best*, but is certainly easiest
14:22.15b00mer_blitzrage: what is the best?
14:22.21blitzrageb00mer_: whatever works
14:22.25b00mer_:)
14:22.26b00mer_ok
14:22.42blitzrageb00mer_: "best" questions are full of variables -- nothing *is* the best
14:22.55Ahrimanesblitzrage: will a "simple" solution like the one app_voicemail uses (sending out via sendmail) do?
14:23.09blitzrageAhrimanes: yah, for sure
14:23.32Ahrimanesblitzrage: ok, should be a simple job then :)
14:23.36blitzrageAhrimanes: nice
14:24.03Ahrimanesblitzrage: you'll be irc'ing over the weekend?
14:24.13blitzrageAhrimanes: oh yah -- I'll be here all weekend working on stuff
14:24.46Ahrimanesblitzrage: ok, i'll poke you sometime saturday or sunday then :)
14:24.55blitzrageAhrimanes: coolio
14:24.58*** join/#asterisk mut (n=animenod@65.111.222.120)
14:25.23htimshello, when i try to call someone via sip, the called phone rings only about 20 secounds. can somone please help me to find and maybe solve this problem?
14:25.46blitzragehrmmmm... whats a good way of setting and checking the return status from a macro
14:26.17Ariel_NoOp
14:26.37*** join/#asterisk saftsack (n=saftsack@p54A7E42E.dip.t-dialin.net)
14:26.48*** join/#asterisk X-Gen (n=x-gen@dsl-145-223-137.telkomadsl.co.za)
14:26.59Flautoi setup wakeup call agi for my asterisk, but when i tried to setup a time, it rings right away, it seems that the system does not have the sense of setting up a future time.
14:28.56Ahrimaneshtims: probably the Dial() command in your dialplan has a 20 second timeout set
14:30.12htimsAhrimanes: no.. i tried the dial command with and without a timeout value...
14:30.46*** join/#asterisk T-Squared (n=T-Square@hidden.serreyn.com)
14:31.08*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
14:31.17Ahrimaneshtims: hm strange
14:31.36T-Squaredok what is the trick to get a sip phone to show the 2nd call coming in on another line
14:31.55ManxPowerT-Squared, That would be specific to each specific model of phone.
14:32.23ManxPowerThe first step is usually turning off call waiting.
14:32.28T-Squaredpolycom 501/601
14:32.49ManxPowerAh.  That's easy.  Set 1 call per line in the polycom config.
14:32.59htimsAhrimanes: i also removed everything from my sip.conf except the configuration for 2 sip providers...
14:33.22Ahrimaneshtims: hm dont think i can help here unfortunately
14:33.40T-Squaredok done that, but my phone does not ring other line when I am on it (asterisk sees phone as busy?)
14:36.26drfoomod2T-Squared: when you get it to work, can you add the procedure to http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#722Polycom
14:36.45drfoomod2T-Squared: i'm going to use the same phones on *, and I'd to have the same functionality
14:36.49puzzledwhen I call from analog phone A via port 1 on a tdm400p to analog phone B on port 2 and pickup phone B I hear the dialled number in dtmf tones. Anyone know how I can make the dtmf go away?
14:39.58tzangerpuzzled: don't dial with digits
14:40.00T-SquaredI had it working before
14:40.10tzangerpuzzled: Dial(Zap/2/,,)
14:40.23drfoomod2T-Squared: what changed?
14:40.30tzangerer sorry just Dial(Zap/2)
14:41.33*** join/#asterisk RoyK (n=roy@84.210.23.221)
14:41.44ManxPowerT-Squared, we have each line register as a different sip username, the let asterisk reroute the call when it gets a BUSY
14:42.43ManxPower99% of the time when dialing an FXS port you do NOT put a number to dial in the Dial() line.
14:42.56T-Squaredhad to rebuild my asterisk@home box, sure wish they would do upgrades, I really can't teach my assistant dial plans, too technical for him
14:43.24ManxPowerT-Squared, Too bad we don't know anything about Asterisk@Home/FreePBX/AMP here.
14:43.48T-Squaredyada yada yada and thanks for all the fish, I don't want to hear it
14:44.25T-Squaredthat's like saying we only know about engines and don't drive cars here
14:44.29ManxPowerT-Squared, pretty much anything we tell you will be wrong because it AMP/A@H will do it differently.
14:44.48NuggetManxPower is absolutely correct.
14:44.51T-Squaredwe're talking technologies, theories, not implementations
14:45.03ManxPowerT-Squared, No, it's like saying we work on gasoline powered cars, but you are bringing in a hydrogen powered car to be repaired.
14:45.27T-Squaredno I'm talking about driving, you drive them both yes?
14:45.42ManxPowerMaybe.
14:45.43puzzledtzanger: thanks! that worked like a charm.
14:45.57puzzlednow to find why the flash button on the phone doesn't do anything
14:46.01T-Squaredthey both have engines that move the tires around and around and it goes right?
14:46.04ManxPowerBut you are not talking about dirving the cars (dialing calls) you are trying to modify your car.
14:46.41*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
14:46.45Ariel_T-Squared, do you have each line registered on only the first line
14:46.52coppicesounds like everyone's talking twaddle :-\
14:46.56T-SquaredI'm not arguing the point, I asked a question in how to do it in asterisk, and what settings were required, I will take care of the particular implementation differences
14:47.33ManxPowerT-Squared, *shrug* turn off call waiting on the polycom, modify your Dial macro to send the call to the 2nd line when the first line is busy.
14:47.59ManxPowerand you MAY have to register each line as a different SIP userid.
14:48.02*** join/#asterisk saftsack (n=saftsack@p54A7E42E.dip.t-dialin.net)
14:48.41Ariel_T-Squared, if you have each line setup via freepbx on the polycom you can create a ring group of one then, to next one. Like a roundrobin
14:49.05MikeJ[Laptop]http://uploads.ungrounded.net/content.php?id=7&name=7_civic.swf&title=Souped-up%20Civic&date=1145592000&quality=b&uj=0&w=400&h=200
14:51.19T-SquaredI thought about doing it with mulitple lines, but I had this working with one line, I'm going to go and just review my config compared to the original one
14:51.54DrukenManxPower: does asterisk allow multipul registrations yet?
14:54.46*** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx)
14:55.16drfoomod2getting back to the engine/car analogy for a second
14:55.46drfoomod2it's like having the same Catepillar engine in a bulldover and a Ford F650
14:55.58blitzrageT-Squared: not all cars are the same
14:56.26drfoomod2same _engine_, different implementation
14:56.31coppicetrue. some are much dirtier than others
14:56.39Greek-Boyso there is no way to phone a skype user from asterisk?
14:57.07blitzrageGreek-Boy: the answer is still no
14:57.22coppicedial out from * to the PSTN, then into a skype PSTN gateway and you're home :-)
14:57.35blitzragecoppice: lol -- thats what I was thinking :)
14:58.07Drukenskype is done by alpha-numeric user name isn't it?
14:58.17ManxPowerDruken, NO!  However Polycom and Cisco phones will only register ONCE if you configure multiple lines as the same user/secret
14:59.17DrukenManxPower: ok, i was just asking, i have multipul locations for my extension :) i'd make my life alot easier with multipul registrations
14:59.29*** join/#asterisk zeppelin_ (n=zeppelin@201.15.175.147)
15:01.19*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
15:03.08*** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it)
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15:08.58coppicejust who would buy anything from a site like Astcode.com? it exudes dodgy :-\
15:09.52htimsquit
15:12.40cpmtelemarketers?
15:13.51ManxPowercpm, no need to be vulgar
15:14.10*** join/#asterisk itunali (n=ibrahim@88.247.84.55)
15:14.19cpmsorry
15:14.44drfoomod2coppice: well for one, their English sucks
15:15.12mutpoor telnet
15:15.16mutthey have no idea whats going on
15:15.17*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
15:15.30*** join/#asterisk vittorio (n=andreas@cm113-144.liwest.at)
15:17.21vittoriohi, is there an irc channel for pbx4linux available?
15:18.14BladeRunner05Hi all, I have a problem: i'm using asterisk 1.2.7.1 with chan-capi-vm.0.6.5, in my dialplan I have a menu and when the caller press the key it execute exten => 104,1,Dial(SIP/xxxx,20,r)
15:18.30BladeRunner05but the caller don't hear nothing while it ring to xxxx
15:18.46BladeRunner05and asterisk say: chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS
15:18.49*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
15:19.06BladeRunner05I also try to remove ,r but is the same
15:21.52*** join/#asterisk `Kevin (n=Kevin@64.243.236.10)
15:22.24`KevinI am using a TE110P card, and cannot get past the error of..
15:22.26`KevinZT_SPANCONFIG failed on span 1: Invalid argument (22)
15:23.32*** join/#asterisk copantl (n=galellop@63.245.93.138)
15:23.38copantlhi
15:23.48copantlany body use oh323?
15:24.21jsharp`Kevin:  What does your zaptel.conf look like?  Put a copy of it on pastebin.ca
15:24.49copantli need help with oh323.conf
15:27.09`Kevinjsharp: http://pastebin.ca/50428
15:28.14mmlj4this is a must-read: http://www.theregister.co.uk/2006/01/27/bofh_2006_episdoe_4/
15:28.23*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
15:29.17copantlwhy this? :Apr 20 03:49:19 WARNING[29805] chan_oh323.c: Inbound call 'ip$XX.XX.XX.XX:XXXXX/25695-8b9f4a69' dropped due to call-limit violation
15:29.42`Kevinjsharp: I also have a TDM22B installed, but not yet configured
15:29.50jsharp`Kevin:  There's your problem.
15:30.00`Kevinjsharp: either configure or remove it?
15:30.14jsharpOr make sure the TE110 driver loads first.
15:30.19`Kevinjsharp: k
15:30.30jsharpIts seeing the ports on the TDM22B as "span 1"
15:31.01`Kevinmodprobe wctdm wcte11xp
15:31.02`Kevin?
15:31.35jsharpmodprobe wcte11xp, then wctdm
15:32.00*** join/#asterisk bkw__ (n=bkw_@ppp-70-128-114-218.dsl.tulsok.swbell.net)
15:32.01`Kevinah
15:32.03`Kevinworks ty so much :)
15:33.22ManxPowercopantl, so what is the problem?
15:33.42copantlwhy this? :Apr 20 03:49:19 WARNING[29805] chan_oh323.c: Inbound call 'ip$XX.XX.XX.XX:XXXXX/25695-8b9f4a69' dropped due to call-limit violation
15:33.55ManxPowercopantl, becasue you have a calllimit set.
15:34.29ManxPowerremove the call limit and the error should go away.
15:34.36copantlafter of that my asterisk is folling down
15:34.51copantlin oh323?
15:34.58copantlright?
15:35.01ManxPowercopantl, I would assume so.
15:35.12copantlok
15:35.13ManxPowerYou would have had to set it, it doesn't get enabled by default
15:35.39ManxPowerYou're not using AMP/FreePBX/Asterisk@Home, are you?
15:36.05ManxPowerBladeRunner05, you do not have an /etc/asterisk/indications.conf
15:36.19ManxPower"r" NEVER fixes lack of ringing.
15:36.22copantlnop
15:36.29copantlim not using amp
15:36.39copantlsorry is not with me
15:36.56copantlthe call limit in my oh323 is comented!!
15:37.22copantli guess is taking from default
15:37.29websae<PROTECTED>
15:37.39ManxPowerthe default is to not have call limits
15:37.43X-Genoooh websae wants to get personal
15:37.49websaeyep
15:37.50websaelol
15:38.46copantli don't understand... is unlimited right
15:38.56*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
15:38.56copantlwhy the message?
15:39.38copantl;
15:39.39copantl;crlCallNumber=20
15:39.39copantl;crlCallTime=20000
15:39.39copantl;crlThreshold=30
15:39.39copantl;
15:40.13*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
15:40.26copantlthis means a accept a call in 1 second
15:40.32copantl??
15:40.51*** part/#asterisk itunali (n=ibrahim@88.247.84.55)
15:41.01Drukeni r bored....
15:41.15Drukeni hate days at home... nothing todo...
15:41.44*** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx)
15:41.45ManxPowercopantl, not many people use H323 with Asterisk
15:41.49*** join/#asterisk itunali (n=ibrahim@88.247.84.55)
15:41.58*** part/#asterisk itunali (n=ibrahim@88.247.84.55)
15:42.26coppicebecause SIP is just sooooo much better :-)
15:42.31websaeWe us H323---however it was quite a hassel to get going
15:42.31ManxPowergreat, I'm short about 1,000 ft of coax
15:43.33websae1000ft short...hrm that's not good
15:43.34*** join/#asterisk sumerik (i=sumerik@mastermold.lodden.com)
15:43.43DrukenManxPower: for what ?
15:43.44*** join/#asterisk bigblock02 (i=pplwro@ool-43504a4a.dyn.optonline.net)
15:43.56ManxPowerDruken, wireing the campground
15:44.08Drukenwebsae: i'd rather be 1000 feet short then like 10...
15:44.09Drukenhehe
15:44.13coppicejust pull it a bit tighter round the corners and you'll be home and dry
15:45.23Drukenoh fucken great
15:45.30Drukenit's supposed to rain all weekend
15:45.40ManxPowernow I need to size the conduit for 2xRG-6, 3xCat5, and 4x4-strand miltimode fiber
15:46.45jsharp4"
15:47.36coppicei think it will be dry all weekend. well, dry from rain, but probably somewhat sweaty
15:48.13coppicemultimode fibre is for paupers. real men only use monomode
15:48.46ManxPowercoppice, I'm a pauper
15:48.52*** join/#asterisk jofre (n=jofre@200.180.68.159)
15:49.26Drukencoppice: well, the weather network says rain all weekend... starting tonight :(
15:49.40coppicemultimode is a dead end. at least one you lay monomode you're good for at least 320G per fibre
15:50.03coppiceDruken: Nope. the weather forecast says hot and dry
15:50.36Drukenfor where you are :) smart ass
15:51.40SupaplexYou're both right, however the forecast says it will be mild and partly cloudy
15:51.53*** join/#asterisk apardo (n=apardo@241.Red-80-34-202.staticIP.rima-tde.net)
15:52.11Ariel_Hot and sticky humid weather here.
15:52.38coppiceAriel_: at least one of you got it right
15:53.01Ariel_~weather KTMB
15:53.39coppice~weather HK
15:54.41Vagabondwhy do they even bother with miami weather forecasting? there's like 3 options: hot and sticky, rain and hurricane
15:55.03drfoomod2~weather KDIX
15:55.14Ariel_Vagabond, well it's hot outside and actually I don't see any clouds at all clear blue skys.
15:55.25*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
15:55.26coppice~weather 45031
15:55.44coppice~weather VHHH
15:55.57coppicehurray
15:56.12Ariel_coppice, it's cooler there then here.
15:56.29coppiceit is night time
15:56.39Katty:>
15:56.41coppiceit was hot at midday
15:56.44Ariel_coppice, forgot that
15:56.50*** join/#asterisk _DAW (n=bob@adsl-150-58-174.msy.bellsouth.net)
15:56.54*** part/#asterisk _DAW (n=bob@adsl-150-58-174.msy.bellsouth.net)
15:57.28*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
15:57.29mut~weather K3RN
15:57.36mutpf
15:57.37mutghetto
15:57.44*** join/#asterisk bkw__ (n=bkw_@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net)
15:57.46mut~weather K55D
15:57.47*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
15:57.51coppice78% humidity == very dry day :-)
15:58.00mutnice weather
15:58.06mut~weather KGOV
15:58.13mutyay
15:58.21Vagabond~weather KMSS
15:58.57MikeJ[Laptop]~weather KPTK
15:59.01Ariel_mut, its so dry there
15:59.05Juggie~weather moo
15:59.14mutactually it's quite nice
15:59.23mutspecailly sittin outside
15:59.34mutnot too windy, nice and cool
15:59.39muthot sun shinin down
15:59.44MikeJ[Laptop]mut, indeed
15:59.49Ariel_we have a very dry day for our standards here it's down to 53% H here.
15:59.55Juggie~weather CYOW
16:00.25MikeJ[Laptop]ok.. so who had to actually look up their codes, and who arlready knew them
16:00.50MikeJ[Laptop]you fly?
16:00.53mutonly one i knew was
16:00.56mut~KCMX
16:00.59mutwhere i went to school
16:01.03mut~weather KCMX
16:01.12coppiceyeah, like VHHH was obviously HKIA :-)
16:01.17MikeJ[Laptop]I don't fly nearly as much as I would like to
16:01.28muti';ve never flown o_O
16:01.36MikeJ[Laptop]you should
16:01.47MikeJ[Laptop]or move the hell out of greyling
16:01.48ManxPowerIt's faster than walking
16:01.49mutsure, soon as i get rich
16:01.51coppiceif you actually want to fly, i guess you don't do it too often
16:01.53Ariel_upgrading our airplane from 150hp to 160 so we can save gas and carry more load
16:01.55mutactually i live in lupton
16:01.56mutheh
16:01.59BladeRunner05I'm using asterisk 1.2.7.1 with chan-capi-vm.0.6.5, in my dialplan I have a menu and when the caller press the key it execute exten => 104,1,Dial(SIP/xxxx,20,r) but the caller don't hear nothing while it ring to xxxx and asterisk say: chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS I also try to remove ,r but is the same
16:02.00mutgayling is the closest place
16:02.04MikeJ[Laptop]mut, it's not "that" expensive
16:02.07mutgrayling
16:02.10cpmheh
16:02.10MikeJ[Laptop]y
16:02.20mutyes it is..
16:02.25MikeJ[Laptop]I drive thrgouh going up north
16:02.38*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
16:02.39mutused to live in mio
16:02.39MikeJ[Laptop]well.. gas is more now I suppose.
16:02.42mutwork in rose city
16:02.45coppicelast week I had a whole day in Hangzhou airport staring at a broken plane. today I had ages stuck in nanjing airport because of air congestion. the jet set life is *such* fun
16:02.50*** part/#asterisk ckruetze (n=ckruetze@62.214.75.2)
16:02.53mutprobly drive thru both those
16:02.54Kattymut: i live in the city of roses.
16:03.00MikeJ[Laptop]rose city... heh... you know jim wauldron?
16:03.12MikeJ[Laptop]bad frog brewery guy
16:03.18mutused to be a customer of ours
16:03.21MikeJ[Laptop]heh
16:03.25mutdunno him personally tho
16:03.35MikeJ[Laptop]they had a resturaunt down here for a while
16:03.43mutyea
16:03.45MikeJ[Laptop]he's my step mothers cousin
16:04.02Ariel_coppice, don't fly much in jets... mostly small GA airplanes...
16:04.09*** part/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk)
16:04.22MikeJ[Laptop]I don't like to fly without being able to see out the front
16:04.38coppicedon't fly much in prop planes.. mostly A320s and up :-)
16:04.57Bert-bye here nice week end to all ;)
16:05.04zoacoppice mainly owns 747's :)
16:05.06fourcheezeit seems there's a cool date coming up next month:  01:02:03 04/05/06.
16:05.24*** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net)
16:05.29coppiceI don't like spending hours in airports and small metal boxes
16:05.39muteh well
16:05.46muti've no where to fly anyway
16:05.48mutso i don't fly
16:05.56muti never even been outta michigan
16:06.12Dandan?
16:06.13fourcheezehey, I've never even been there
16:06.15MikeJ[Laptop]mut, the best place to fly in MI is the jackson airport
16:06.15Dandanmut: why?
16:06.19Dandanany reasonable fear?
16:06.22Ariel_fourcheeze, that has passed already  april 5th 2006
16:06.28coppicemut: i was like that till my mid 30s. i've unfortunately been compensating since :-)
16:06.31mutcause i have no where to do..
16:06.35fourcheezeAriel_: only if you do your date in the wrong order
16:06.39mutto go*
16:06.54muti make enough to live not, not enough to have fun
16:06.59mutto live on*
16:07.01MikeJ[Laptop]hehe'
16:07.02mutman i can't type today
16:07.07Dandani c :)
16:07.10coppiceI've been to michigan. its one of the few places a really wanted to get out of.
16:07.12Ariel_fourcheeze, I know it as M/D/YYYY
16:07.17mutheh
16:07.20muty
16:07.22MikeJ[Laptop]coppice, join the club
16:07.23Dandanbut why you do not fly? that i can't comprehend
16:07.27Dandani love flying :)
16:07.30fourcheezeAriel_: what kind of crazy place uses a middle-endian order?
16:07.41mutwhere should i fly to?
16:07.46Dandani do not know...
16:07.49Dandananywhere
16:07.50mutexactly
16:07.50ronnhi
16:07.51MikeJ[Laptop]jackson MI airport
16:07.51Ariel_stay away from Arkansas
16:07.59Dandanforeign countries have so much to offer
16:08.08MikeJ[Laptop]they have a great greasy spoon resturaunt there
16:08.10ronnwhats this
16:08.21fourcheezeAriel_: we have D/M/Y
16:08.21muttake me 4 years to save money for a trip somewhere else
16:08.25coppicewhat is there to love about being in an uncomfortable cramped seat, with lousy food and entertainment for hours?
16:08.51ronnhow old are you ppl?
16:08.59Ariel_fourcheeze, nope you must not live in the US
16:09.07mut22.5
16:09.08Dandanronn: almost chris's age
16:09.09Dandan:D
16:09.13fourcheezeAriel_: no shit
16:09.42muti prefer y-m-d
16:09.47mutor d-m-y
16:09.50Ariel_looks like allot of young folks here
16:09.56Dandan*christ's age
16:10.03DandanMan, I Can't type today...
16:10.03Dandan:D
16:10.04nahirean<-- not young.. 24 years old.. im a gezzer :)
16:10.10nahireangeezer, even
16:10.22fourcheezeDandan: 2000?
16:10.25Ariel_your all young enough to be my kids
16:10.28Dandannahirean: no worries, I am almost 30 :)
16:10.37coppiceeveryone should use ymd. its the international standard
16:10.40nahireanDandan: Does it ever catch up?  I still feel 16 ;)
16:10.52coppicejust like the US is officially metric :-)
16:11.02*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
16:11.07Dandanalmost 30 :))
16:11.20Ariel_metric hummm not around here.
16:11.28Dandanhere (in the US) we count years by feet!
16:11.29Dandan:)
16:11.32Dandannot in meters!
16:12.13fourcheezeit's about time the US went metric
16:12.20mutso does that make a form owner ancient?
16:12.21Drukenwtf?
16:12.22mutfarm
16:12.25Ariel_funny thing is most of the items here in the US now have both of them on the label
16:12.29coppicethe US went metric over a century ago
16:12.29*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
16:12.31Drukenuhmm... i don't think the us would go metric
16:12.44coppiceand several times repeated going metric since then
16:12.50Dandanfourcheeze: it is getting there...
16:12.51Ariel_coppice, but we don't use it much
16:12.58terrapenanybody played with zttool
16:13.00coppicesomehow it doesn't actually stick
16:13.03Dandanfor example hospitals already measure newborns in cm's and kg's
16:13.07terrapenthe damn thing doesn't render correctly in my terminal
16:13.09fourcheezethe UK only has road distances left to go
16:13.17MikeJ[Laptop]the auto industry keeps the US from going metric for real
16:13.26fourcheezecrazy to buy fuel in litres and travel 200 miles on it
16:13.30MikeJ[Laptop]everyone would need to get new tools
16:13.33coppicedo UK cars now come with speedos in km/h?
16:13.39MikeJ[Laptop]not that they don't have them now..
16:13.47fourcheezeall UK cars have both marked
16:13.50MikeJ[Laptop]the issue is the metric sized bolts
16:13.58fourcheezeunless they are digital
16:14.00MikeJ[Laptop]not the spedos
16:14.04fourcheezeI presume the digital ones are switchable
16:14.07*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
16:14.14MikeJ[Laptop]most in the US have both onthe spedos.
16:14.25MikeJ[Laptop]the streetsigns on the other hand are mostly miles
16:14.38Dandantheir speedos are in yards/second
16:14.38Dandan:)
16:14.38fourcheezethere's pressure here to ditch miles
16:14.39coppiceall the japanese cars built in the US are metric. aren't they almost the majority of the US car industry now? :-)
16:14.55Ariel_gallon for gas, what if we had to pay for it in liters
16:14.56MikeJ[Laptop]heh
16:15.01fourcheezecoppice: I wouldn't be surprised
16:15.02jsharpfurlongs per fortnight
16:15.07fourcheezeyou'll end up going metric
16:15.10mutheh
16:15.14mutif we started paying per liter
16:15.16MikeJ[Laptop]well... more or less.. chrysler is not techniceally an american company..
16:15.16mutpeople would go nuts
16:15.17fourcheezebecause it is inexorable
16:15.21MikeJ[Laptop]so it's just GM and ford
16:15.23mut"OMG GAS IS SO CHEAP NOW!"
16:15.43Ariel_cheap
16:15.49docelm0mut your on crack..  its 3 buks in tampa
16:15.58terrapenBut what about the McDonalds Quarter Pounder?  And NWA's "100 Miles and Runnin'"?
16:15.59fourcheezeI guess your maths exams must be harder than ours
16:16.00MikeJ[Laptop]docelm0, read what he wrote
16:16.00nahireanpeak oil is upon us.. gas will not last ;)
16:16.02terrapenwe can't change now!
16:16.03Ariel_regular is $ 3.04 per gal here
16:16.20b00mer_anybody run chan_sccp in a production environment?
16:16.20Vagabondits $3.14 here
16:16.24docelm0where is here?
16:16.27fourcheezedocelm0: $3 a gallon is cheap
16:16.30nahirean$2.77 here
16:16.34fourcheezeisn't the US gallon different to the UK one ?
16:16.34VagabondUpstate NY
16:16.35terrapen$2.35 here in Utah
16:16.44docelm0I am going to lunch..  I will tell you the price when I Get back
16:16.45nahirean2.77 in jwoisey
16:16.47terrapenoh, and we have great skiing, too
16:16.51b00mer_US gallon = UK gallon
16:16.51Ariel_fourcheeze, yes
16:16.56BladeRunner05I'm using asterisk 1.2.7.1 with chan-capi-vm.0.6.5, in my dialplan I have a menu and when the caller press the key it execute exten => 104,1,Dial(SIP/xxxx,20,r) but the caller don't hear nothing while it ring to xxxx and asterisk say: chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS I also try to remove ,r but is the same
16:16.56coppiceyou can buy all sorts of stuff in china in imperial units, even though they have never really been a user of imperial units. weird
16:17.03docelm0I thought UK was metric and didnt use gallon
16:17.09b00mer_US gallon != UK Litre
16:17.20fourcheezedocelm0: we don't use the gallon much
16:17.29fourcheezeat least not on pumps
16:17.34fourcheezehowever
16:17.46bigblock02roughly 3L per gallon...
16:17.49fourcheezewe currently pay £1 for a litre
16:18.04fourcheezeand £1 ~ $1.77 I think
16:18.05Ariel_wow about 5 dollars per gal there then
16:18.13coppiceIn India they had so much trouble making metric units stick, they outlawed the old ones :-). It is illegal to sell things like a thermometer capable of reading in F
16:18.13Vagabondwhen I was in Ireland it was like 2 euro a litre
16:18.16DandanHm, overnight it went up to $3.05/gal here
16:18.33*** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net)
16:18.45Vagabondit jumped 10 cents overnight here, dunno what it'll be by the end of today
16:18.52terrapenMy truck uses Premium Unleaded and gets about 12mpg, MAX
16:19.06docelm0I get 36 MPG..
16:19.11Ariel_my truck uses reg. and gets 22 MPG
16:19.11bigblock02what kinda truck?
16:19.12docelm0I love it.. I fill up like every 2 weeks
16:19.22terrapenLand Rover Defender 90
16:19.26coppicei remember when gas was $3 a gallon in the UK, but only because I'm older than most of you :-)
16:19.31fourcheezeif the mass of a cubic metre of water is 1000KG, what is the mass of a cubic foot ?
16:19.32jsharpIts funny that the cost per gallon can jump 2 or 3 times in a day, but the cost of the fuel in the underground tanks didn't change at all.
16:19.32distortionmmm my hemi and its 12mpg
16:19.34bigblock02must be high compression, supercharged, or something like that to REQUIRE 93/94
16:19.35MikeJ[Laptop]I work from home, I fill up my gas tank once a month
16:19.48bigblock02or have the timing bumped way up
16:19.52terrapenyep
16:19.55Ariel_MikeJ[Laptop], same here
16:20.14nahireani am lucky to live close to work, i fill up one every 2 weeks
16:20.18bigblock02most "fancy" cars say they need premium when they for the msot part run just fine with 87
16:20.18terrapenand right now, it's running really rich and is getting around 8-10 mpg
16:20.18nahireanonce*
16:20.19MikeJ[Laptop]fourcheeze, you run on gas?
16:20.26terrapeni need a new throttle position sensor :(
16:20.34bigblock02you are only preventing detonation with octane...
16:20.37terrapenyep
16:20.38fourcheezeMikeJ[Laptop]: diesel
16:21.11bigblock02i figure 2mpg tops on the quarter mile strip...
16:21.16Mikecoppice, are indications.conf for br the same for mx?
16:21.17MikeJ[Laptop]what the hell burns 8 gal/hr
16:21.25Ariel_MikeJ[Laptop], my plane
16:21.41justinu|laptop8 gph isn't bad
16:22.03jsharpTakes 100LL?
16:22.03coppiceMike: I don't think so. you can look in my XML file of all the world's indications
16:22.07tier_1snapple
16:22.09Ariel_we are upgrading the engine and should get 10 more hp and get down to about 7.5 per hour..
16:22.25Mikecoppice, where can i find it?
16:22.33coppicein spandsp
16:22.43MikeJ[Laptop]Ariel_, 15 1/2 gal/hr :(
16:22.43Mikecoppice, ok
16:22.49Ariel_coppice, did you get your web back up?
16:22.51*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
16:22.53coppice8gph isn't much.
16:22.56nahireanhttp://www.dslreports.com/forum/remark,15941943
16:23.08coppiceAriel_: you mean soft-switch.org?
16:23.16Ariel_yes
16:23.35coppicesome nice people in .mx had an up to date mirror of the site :-)
16:23.40justinu|laptopso nufone is gone?
16:23.55Ariel_nice
16:24.01Ariel_justinu|laptop, no
16:24.16docelm0No they arents..  I am providing some of their services
16:24.22justinu|laptopoh
16:24.23fourcheezecoppice: it depends how far you can go in your hour
16:24.46coppicelike Linus Torvalds said, backup is for wimps. real men put anything important on the internet, and let the world mirror it :-)
16:24.47justinu|laptopsr22 burns something like 20gph, but its faast :)
16:24.54docelm0Termination is working.  Trying to get TF back online ASAP but thats out of my hands until the upper powers make some decisions
16:24.58Ariel_fourcheeze, yep... 125mph if no tail wind.
16:25.02*** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
16:25.19Ariel_justinu|laptop, I can't afford an SR22
16:25.21mostyi want to record a call (in+out) that is being dialled out via a sip provider, how can i do that?
16:25.21fourcheezeAriel_: still only 15 mpg
16:25.26coppicefourcheeze: an oil tanker doesn't go far in an hour, but burns quite a lot of fuel :-)
16:25.33Ariel_fourcheeze, about right
16:25.34jsharpmosty: App_monitor
16:25.35fourcheezebut I suppose you get to fly in a straight line
16:25.59fourcheezecoppice: well in that example the other factor is how much it holds
16:26.02mostyjsharp: thanks, i'll look it up
16:26.21coppicea 747 burns quite a lot per hour
16:26.21Ariel_well I like that it only takes 1.75 hours to get to orlando from here instead of driving 4 hours.
16:26.31fourcheezeunfortunately aeroplanes are generally very poor economically
16:26.41MikeJ[Laptop]so what happened with nufone... did there tdm get cut off?
16:26.48jsharpholes in the sky you pour money into
16:26.48Ariel_MikeJ[Laptop], no
16:27.03*** join/#asterisk |omni| (i=rob@216.64.178.146)
16:27.17MikeJ[Laptop]so what happened?
16:27.23nahireanAriel_, Why do you say no?  It's not working..
16:27.25Ariel_MikeJ[Laptop], not a thing.
16:27.34Ariel_nahirean, works fine here
16:27.47MikeJ[Laptop]hmmm
16:27.57MikeJ[Laptop]somone jsut said all the toll free's were down
16:28.02Ariel_MikeJ[Laptop], there 800 number service provider wanted to pull the plug on them.
16:28.12nahireanAriel, Looks like they did
16:28.26coppiceI wonder how much the rise in fuel prices will affect air freight
16:29.03mishehucoppice: you'll know as soon as the union boys turn you upside down to shake out your change.
16:29.14*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
16:29.45coppiceits amazing how much freight goes by air
16:31.05*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
16:31.15JonR800indeed.. looks like it's pulled.. my 800 is down as well
16:31.37justinu|laptopwhy would anyone do such a thing to loveable jerjer?
16:31.47tzangerMikeJ[Laptop]: their clec cut them off since they weren't grwoing fast enough apparently.
16:31.57asterboyIs mpg123 suppose to hang around in the process list when there is no activity on *?
16:32.01JonR800poor guy.
16:32.34BladeRunner05I'm using asterisk 1.2.7.1 with chan-capi-vm.0.6.5, in my dialplan I have a menu and when the caller press the key it execute exten => 104,1,Dial(SIP/xxxx,20,r) but the caller don't hear nothing while it ring to xxxx and asterisk say: chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS I also try to remove ,r but is the same
16:32.59asterboydon't hear nothing?
16:33.19asterboyand I thought my grammer was bad.
16:33.31BladeRunner05asterboy: the caller don't hear nothing while it rings to xxx extension
16:33.34terrapenanybody else have a Redfone foneBridge?
16:33.48asterboyterrapen, still having fun with that?
16:34.33mostyjsharp: i'm having trouble locating docs on App_monitor at voip-info.org, can you give me a good term to search for? "App_monitor" only turns up hits for the "asterisk variables" page
16:34.37asterboy~redphone
16:34.39jhiveris anybody any good with SER? I'm stuck on something which I'm sure is simple for someone who happens to know :)
16:34.47asterboy~redfone
16:35.32asterboywonder what the question to anser ratio is in here?
16:35.35asterboys/ns/nsw
16:35.39*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:35.45*** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net)
16:36.04justinu|laptopjhiver: you got 5 minutes :P
16:36.31kuku5Emergency !! My card is not answering calls ( not pri )  I can call out, but calling in doesnt cause the call to be picked up!
16:36.35*** join/#asterisk |omni| (i=cathode@216.64.178.146)
16:37.11mostykuku5: does asterisk see the extension ringing?
16:37.30Ariel_terrapen, I know those units
16:37.40Ariel_kuku5, what card?
16:37.45kuku5analog
16:37.55Ariel_x100p or tdm400p
16:38.09kuku5* doesnt see it ringing
16:38.17justinu|laptopjhiver: about to pack up here...
16:38.31*** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-50.prod-infinitum.com.mx)
16:38.33*** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com)
16:39.06asterboyhey kuku5, can you read?
16:39.20kuku5yes - how do I check which card
16:39.38kuku5but its probably teh tdm
16:39.48jhiverhey
16:39.50jhiversure
16:39.53jhiverI have some time :)
16:40.07asterboycat /proc/interrupts
16:40.08jhiverI can't do failure route with ser
16:40.14asterboycheck for tdm
16:40.16*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
16:40.25jhiverI don't know if I'm using the wrong syntax or what
16:40.31kuku569: 2212558909   IO-APIC-level  wctdm, eth0
16:40.31kuku5177: 2158762403   IO-APIC-level  wctdm
16:40.38asterboygreat
16:40.41kuku5( I have 2 cards
16:40.49asterboyyou have 2 cards, one shares an IRQ
16:40.51asterboybad
16:40.53kuku5ahh
16:40.56kuku5I see
16:41.12asterboyworse its ethernet
16:41.14kuku5If I reboot, it works for about 2-3 weeks, then the same thing happens
16:41.43kuku5What can I do asterboy
16:41.58asterboysolve the share problem first
16:42.07asterboydo you have SIP/IAX devices?
16:42.15kuku5no, cisco phones
16:42.27asterboyya, last I checked thats SIP
16:42.31kuku5:)
16:42.55asterboyso your eth0 is probably causing the problem when your system gets heavy traffic.
16:43.01kuku5ah
16:43.11kuku5How can I force a different IRQ?
16:43.15asterboyeach card takes about 1000 interrupts per second.
16:43.20Qwell[]kuku5: Try another pci slot
16:43.24*** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
16:43.25BladeRunner05asterboy: the caller don't hear nothing while it rings to xxx extension
16:43.27Qwell[]or disable stuff you don't need...like usb
16:43.28BladeRunner05I'm using asterisk 1.2.7.1 with chan-capi-vm.0.6.5, in my dialplan I have a menu and when the caller press the key it execute exten => 104,1,Dial(SIP/xxxx,20,r) but the caller don't hear nothing while it ring to xxxx and asterisk say: chan_capi.c:3951 capi_signal_progress: wrong channel state to signal PROGRESS I also try to remove ,r but is the same
16:43.29kuku5the eth0 is builtin
16:43.37DoktorGreggo into biod diable everything you are not using
16:43.41DoktorGregbios even
16:43.45ManxPowerBladeRunner05, DID you create an /etc/asterisk/indications.conf?
16:43.45kuku5ok
16:43.49asterboydiable com ports if not needed also
16:44.08kuku5ok
16:44.26terrapenasterboy, yeah, still playing with my redfone
16:44.27DoktorGregmobo sound card...
16:44.35terrapenactually, I think my T1 crossover cable is wrong
16:44.38terrapeni need to make a new one
16:44.46BladeRunner05ManxPower: no what is that ?
16:45.02asterboykuku5, here is a good read for ya:
16:45.03DoktorGregquestion... what might i use a soundcard with asterisk for anyhow?
16:45.04asterboyhttp://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
16:45.19Qwell[]DoktorGreg: chan_alsa?
16:45.22asterboyDoctorGreg, Paging
16:45.24ManxPowerBladeRunner05, a sample version is in /path/to/asterisk/configs/indications.conf.sample  It's a file to tell asterisk how to indicate ringing when exiting from an IVR
16:45.32asterboysoftphone
16:45.56BladeRunner05ManxPower: I see that, now try to moddy....
16:46.56Hmmhesaysanyone know if there is any easy way to change the from: field with SER?
16:47.30BladeRunner05ManxPower: I change country= with =it but nothing's changed....
16:49.06ManxPowerBladeRunner05, you did a reload in asterisk CLI after putting the indications.conf.sample (renamed to indications.conf) in /etc/asterisk ?
16:50.14BladeRunner05ManxPower: done, but is the same ...
16:50.55BladeRunner05ManxPower: I think that this depends from chan-capi-vm
16:51.18DoktorGregQwell, yah
16:51.27DoktorGregoh
16:51.34DoktorGregasterboy already answered me
16:52.07*** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se)
16:52.07*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
16:52.46fndudein indications.conf, is it possible for me to set callwaiting=no, or to adjust the db levels of this tone in some way?
16:54.04ManxPowerfndude, no, that is done in OTHER files.
16:54.09ManxPowerlike zapata.conf, for example
16:54.58fndudelike sip.conf..
16:55.53fndudeI have users that are complaining about the volume of this tone. I assumed it was being generated by the phone, then I found these tone options.
16:56.20ManxPowerfndude, ALL that stuff is in the phone configuration file
16:56.27*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
16:57.05*** join/#asterisk jofre_ (n=jofre@200.180.68.159)
17:00.13Hmmhesaysany brits in here right now?
17:00.29asterboyCan someone check to see if mpg123 is suppose to hang around in the process list?
17:00.39Qwell[]asterboy: if?
17:00.42[TK]D-Fenderoops :)
17:01.14fndudeok thanks.
17:01.26Hmmhesayshttp://www.engadget.com/2006/04/20/uk-drivers-trust-gps-more-than-their-own-eyes/
17:01.38asterboyWhen there is no * activity there are 2 mpg123 process still around.
17:01.51BladeRunner05ManxPower: no other ideas ?
17:02.21ManxPowerBladeRunner05, If I had other ideas I would have said something.
17:02.34asterboyroot     12877  2621  0 09:43 ?        00:00:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 Dusty
17:02.46BladeRunner05ManxPower: tnx
17:03.15asterboyQwell[], if as in condition, or if as in acronym IF?
17:03.29ManxPowerIf a company wants my business it would be nice if I didn't have to enter my credit card info in 4 times and then get an error saying the web site is down.
17:04.01mutanyone know why only ppl calling from nextel cell phones don't hear a ring on their end
17:04.06mutbut it's running on the voip customers end
17:04.08[TK]D-FenderManxPower : Thats so they can reconnect you each of their identity-theft partners before giving you THEIR rejection ;)
17:04.18mutcall works, theres just no ringing generated by nextel
17:04.24mutwhat're some causes of that
17:04.31ManxPower[TK]D-Fender, LOL!  No this is a legit company.
17:05.04[TK]D-FenderManxPower : But are you on the site you THINK you are!
17:05.22Hmmhesaysyeah , thats right
17:05.23asterboythey are a legit identity-theft company
17:05.23ManxPowerI picked "new customer" entered in all my info, then was told an account with that name is already in the system, then I logged in, entered in all my info again only to be told that I have to go into the account setup and change the billing address when I moved, do I did that, then was told the site was not available when I went to check out.
17:05.38ManxPoweremasco.com
17:05.41ManxPower..
17:05.44ManxPowerenasco.com
17:06.13asterboyTK, plant those next to some Cannabis Sativus
17:06.14*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
17:06.42*** join/#asterisk vlrk (n=vlrk@202.65.134.119)
17:06.59docelm0oi!
17:07.19mutso no one has run into that issue before i take it?
17:07.33docelm0No ring from endpoint?
17:07.38docelm0progress issue
17:07.48docelm0probably beyond you
17:07.49*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
17:08.12mutbeyond me
17:08.23mutwhy do all other telco's work fine then
17:08.42mutit's sending the same progress indication as it would any other
17:08.42docelm0Who are you terminating to?
17:09.04mutterminating which end
17:09.09*** join/#asterisk tzafrir_home (n=morphing@bzq-179-75-202.cust.bezeqint.net)
17:09.14mutvoip -> ast -> cisco -> pri
17:09.16docelm0Asterisk might be..  but the last mile might have issues sending progress via the network you chose
17:09.21docelm0what kinda cisco?
17:09.24mut5350
17:09.34docelm0ohh..  on your dialpeer add the following:
17:09.39docelm0simple fix
17:10.17BladeRunner05ManxPower: I resolve, it depends by capi.conf, now the caller hear a ring, there is a way to change the ring tone with a music ?
17:10.28docelm0<PROTECTED>
17:10.29docelm0<PROTECTED>
17:10.29docelm0<PROTECTED>
17:10.34docelm0That's all you need
17:10.42docelm0Im a cisco guru
17:10.55docelm0and I have 5 5350's...  :)
17:11.02mutheh
17:11.04docelm0put it in your POTs dialpeer
17:11.09docelm0err voip..  hold on
17:11.14mutvoip
17:11.24mutwhat about outgoing CID
17:11.27docelm0Nope.. POTS
17:11.31muti'm sending it to the box but the box just ignores it
17:11.32docelm0its isdn pri?
17:11.49mutis there another?
17:11.55docelm0yes.. CAS
17:11.58mutheh
17:12.02mutrhetoriacal
17:12.27mutsec
17:12.33docelm0anywho..   um, you have to make sure your provider will accept the # your presenting when its sent out ISDN..   if they dont accept it it will not be sent
17:12.44mutyea they do accept it
17:12.44Netgeekswhat is the easiest method to find out what asterisk is dying on if it's seg faulting and core dumping when you start it up?
17:12.46BladeRunner05May I change the ring tone heared by caller with another music ?
17:12.49docelm0I have a couple voice DS3's and TONS of E1/T1's
17:12.52muti put my stupid digium card in there
17:12.58justinu|laptopnetgeeks: run it with -cvvvv
17:13.00mutand cid works fine
17:13.02justinu|laptopor more vs
17:13.07Netgeeksjustin: already done
17:13.11mutnot much of an issue tho
17:13.13docelm0BladeRunner05 yes use m in the dialcommand
17:13.13justinu|laptopit doesn't print out any warning/error?
17:13.17ManxPowerBladeRunner05, read the backtrace doc included with the asterisk source code.
17:13.19muti got a sangoma card coming to replace the 5350
17:13.26Netgeeks<PROTECTED>
17:13.27Netgeeks<PROTECTED>
17:13.27Netgeeks<PROTECTED>
17:13.27NetgeeksSegmentation fault (core dumped)
17:13.28BladeRunner05ok try
17:13.34Netgeeksthats the last of it, justinu
17:13.40justinu|laptophmm
17:13.42NetgeeksI have the core
17:13.43docelm0Netgeeks stop loading app_md5.so
17:13.44docelm0:)
17:13.48justinu|laptopfull log indicates nothing either?
17:13.57Netgeeksfull log shows nothing
17:14.07docelm0mut lemme know if that works for you
17:14.09Netgeekslet me give you background
17:14.18justinu|laptopok, then you'll probably have to recompile without optimizations, with -g2
17:14.23justinu|laptopload core file into gdb
17:14.28justinu|laptoptype "bt"
17:14.42mutdocelm0: how come only nextel does that, do others maybe just force it anyway or something?
17:14.49Netgeeksasterisk running on FC4, using service asterisk start in boot to start.. runs for some random amount of time usually less than 48 hours, crashes with no log error
17:14.50justinu|laptopunless you can figure out what module is making it die
17:14.53muti mean if i send busy back it works
17:14.58mutbut ringing won't
17:15.01Netgeekscannot be restarted until system is rebooted
17:15.09Netgeeksany time try to restart, get the core dump
17:15.13justinu|laptophow about unloading/reloading zap modules
17:15.19ManxPowermut, what happens if you run the Ringing app as your first exten line?
17:15.25BladeRunner05docelm0: ok workds but it play the standard MOH, may I specify one ?
17:15.39mutManxPower: it gives double rings or random weird crap
17:15.40ManxPowerNetgeeks, the backtrace document was not helpful?
17:15.49docelm0yes..  check the wiki as I am not well versed on changing MOH
17:16.18Netgeeksbacktrace document?
17:16.25*** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu)
17:16.27docelm0mut I have found nextel to be asses when it comes to ring and shit..   They dont do it for me either..
17:16.31docelm0have you tried other numbers?
17:16.36sevardSo I guess exten => 4000,2,MP3Player(http://207.200.96.229:8030) doesn't work :)
17:16.49mutother numbers?
17:16.53ManxPowerNetgeeks, um, yes the document that explains how to use gdb to generate a backtrace.  Its inclided in the Asterisk source code tree.
17:17.02Netgeeksah, thanks, will look at it
17:17.22*** join/#asterisk biodroid (n=bcm@cust-206-40-173-219.bos-static.gis.net)
17:17.23vlrkdoes any body have idea on provisiong with snom ?
17:18.12ManxPowerNetgeeks, I'll let you in a little secret:  Much of the stuff in the Wiki (and is commonly wrong in the Wiki) is also in documents in the Asterisk source tree (and that source is usually correct)
17:18.32*** join/#asterisk lzhang (n=rjrae@adsl-69-153-6-179.dsl.snantx.swbell.net)
17:18.36_asrsounds as though jerjer needs to pay his billz
17:18.54_asrmuch like the jerky boys skit
17:19.03docelm0He's paid his bills..  the carrier is being a dick
17:19.30lzhanghi, I've got asterisk group ringing several 501's and 1 601, when one of the 501's pick up, the 601 continues to ring even though another phone is already on that call. What could be the issue?
17:20.01*** join/#asterisk frk2 (n=kvirc@202.141.251.102)
17:20.18frk2I blew up my brand new aastra 9133i :(
17:20.21frk2:( :(
17:20.33Netgeeksapp_macro.so is causing the crash
17:20.49frk2voipsupply sent me a 110v adaptor
17:20.53frk2i didnt even notice
17:21.03frk2i dont understand why it blew up the phone also
17:21.40*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
17:21.57asterboyit can still be used
17:22.09asterboy...as a door stop!
17:22.23asterboypaperweight, bookend...etc
17:22.33nahireanwhy would the carrier "be a dick" if they were making $?
17:22.48asterboycause they are a BIG dick
17:23.22frk2asterboyyyyy :(
17:23.27*** join/#asterisk Jacke (i=jacke@valhalla.bofh.org)
17:23.34frk2i dont understand why the wont went stupid
17:23.44frk2i mean usually only the power adaptor should go south
17:23.51asterboydoes it spell burnt?
17:24.07nahireanbut why would a carrier "Be a dick" .. for what reason if they are making cash off the calls?
17:24.30frk2i replaced the adaptor
17:24.41frk2but the phone doesnt respond anymore :(
17:24.47asterboycause like most big telcos I know, they just don't care.
17:24.55sevardgrr i forgot the command to execute an application within extensions.conf
17:25.04frk2System?
17:25.05asterboyfrk2, even with the proper adapter?
17:25.14frk2asterboy
17:25.19frk2they send me a 110v adaptor
17:25.22sevardfrk2: crap, i thought it was exec()
17:25.22frk2i plug it in 220v
17:25.36asterboythat should be ok.
17:25.37frk2so obviously- the adaptor leaves
17:25.42Netgeeksjustinu: on that other issue the other day, where a bad te410p card caused rtp handling apps to not work..  client has digium looking at the card
17:25.49frk2no no - its not a 110-220v adaptor
17:25.49asterboyI mean , it won't work...but it shouldn't fry anything.
17:26.09frk2the adaptor might leave dude
17:26.11frk2anyways
17:26.13asterboyoh, you put a 110 in a 220
17:26.13frk2the adaptor died
17:26.17frk2yes
17:26.26asterboyso, now you just need the proper adatper
17:26.28frk2so i got the adaptor rewound
17:26.34frk2the adaptor now works
17:26.35frk2with 220v
17:26.41frk2but the aastra wont respond :(
17:26.46frk2i dont understand WHYYY
17:26.59frk2the 220v should have ONLY killed the adaptor , right?
17:27.03Ariel_Netgeeks, the problem with the rtp seems to be on any teXXX board not plug in with at least an loop back
17:27.03frk2any EE guy here? :)
17:27.15*** join/#asterisk nagl (n=nagl@86.59.54.237)
17:27.31tzafrir_homedepends on the curcuit, I guess
17:27.45frk2hmmm. true
17:27.55frk2a stupid adaptor could've caused a spike before it died
17:28.00frk2killing the aastra
17:28.15Ariel_yes it could
17:28.28sevardfrk2: Why wouldn't exten => 4000,2,MP3Player(http://207.200.96.229:8030) work?
17:28.47Jackeanyone running 1.2.7.1 on FreeBSD?
17:28.50frk2dude... i dont know if the mp3player app can stream
17:28.52frk2can it?
17:28.55Drukencause streamin moh is a bad idea :)
17:28.59frk2goddamnit
17:29.00sevardmpg123 can stream
17:29.29frk2i just hope that the aastra has a big fuse
17:29.37frk2what kind of modern electronic uses 48v dc anyways?
17:30.04Drukenalot of heavy duty servers
17:30.06Jackefrk2: most of the telco quipement.
17:30.15Jackeequipement.
17:30.28frk2but a IP phone?
17:30.47Jackefrk2: we're talking cisco power brick? :>
17:31.36coppicefrk2: the entire telecoms industry uses -48V
17:32.32docelm0coppice not ALL of it..
17:32.41docelm0Most of my gear is 110 or 220
17:32.45frk2dude thats fine
17:32.59coppicewell, all the wireline stuff, and a large amount of the wireless
17:32.59frk2but i just didnt expect it on a IP phone
17:33.08frk2IP phone is all digital electronic
17:33.29frk2any CMOS stuff would be on 3.3  or 5v internally
17:33.46coppicewhat relevance does that have?
17:33.49Jackefrk2: do they support poe?
17:34.23frk2jacke they do
17:34.35frk2but i dont have a poe switch
17:34.39Jackefrk2: you got your answer ten.
17:34.40Jackethen.
17:34.51Jackefuck, my keyboard is broken or something.
17:35.53Drukentry cleaning it lately?
17:36.00asterboycould that be why mpg123 sits in the process list with no * activity?  Cause its in some kind of stream mode?
17:36.20*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:36.34Drukenmoh can be made to play while no one is on hold
17:37.00asterboythat could be what is going on.
17:37.29terrapenbrb
17:37.30*** part/#asterisk terrapen (n=cjs@166.70.183.108)
17:39.57Jackelet me repeat my question - anyone running asterisk on FreeBSD?
17:40.03Drukencourse, it's probably just the program sitting there, on pause, waiting for someone to be put on holf
17:40.19Drukensome people do, i'm not one of them....
17:40.39frk2Jacke I have a server.. no digium hardware though
17:40.48Jackesome people for sure are, i'm just looking for them ;)
17:40.59Jackefrk2: which bsd, which asterisk?
17:41.30*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
17:42.25frk2freebsd aint my thing.. but i think its 4.x and asterisk 1.2.5. one of the other guys at work maintains it
17:42.53Jackehm. he ported the 1.2.5 himself to freebsd?
17:43.03sevardquestion: mpg123 will throw a stream to stdout, how do i tell asterisk to eat stdin as my mp3?
17:43.10Qwell[]1.2.x should "Just Work" on freebsd
17:43.20frk2i know for sure the Zaptel stuff didnt compile
17:43.27frk2yeah- it wasn't that big of a deal, really
17:43.38frk2hang on
17:43.40frk2its 5.4
17:43.42*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
17:43.43frk2Freebsd
17:43.44frk2and 1.2.5
17:44.39Jackeqwell[]: anything above 1.2.4 from the ports doesn't compile.
17:44.47Qwell[]Don't use ports
17:45.10Jackeqwell[]: doesn't matter - 1.2.5 and above don't compile.
17:45.28Jackefrk2: you may have the patchset for it handy?
17:45.36*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
17:45.37Qwell[]Have you tried grabbing a standard source tarball?
17:45.48Jackeqwell[]: yep.
17:48.20*** join/#asterisk terrapen (n=cjs@166.70.183.108)
17:48.31sevardDangit, can somebody help me?  I can stream a mp3 that I have on my hdd with * but I can't stream a shoutcast stream.  I'd like an extenison that I dial in and it buffers and streams a shoutcast stream.
17:49.38b00mer_anybody use chan_sccp in a production system?
17:51.34frk2jacke i dont
17:51.43frk2as i said am not the freebsd guy
17:52.32Jackefrk2: ok.
17:56.19Drukensevard: its do-able, but i don't reccomend it
17:56.24Drukenlook on the wiki
17:56.24Netgeeksanyone here who can take a look at a backtrace?
17:57.07sevardDruken: I'm looking on the wiki and all I've found is stuff relating to MoH which I've attempted to adapt but have failed so far.
17:57.43*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)
17:58.45jbalcombI have a multitude of complaints from my users about audio cutting out for a second or two from the outside caller. Could someone please provide some thoughts on the problem or direction for troubleshooting?
17:58.49DrukenYou can have asterisk use a streaming source for on-hold music.
17:58.49DrukenMake a directory and put a 0 size file ending in .mp3.
17:58.50DrukenI called my directory: /var/lib/asterisk/mohmp3-empty
17:58.50Drukenin musiconhold.conf, add a line such as:
17:58.50Drukendefault => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/
17:59.11Drukengod damn, if it's not on there, how did i find it in 30 seconds?
18:00.43*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
18:02.07asterboyMy client is saying the GXP-2000 ring volume is not loud enough...anyone have suggestions to boost?
18:02.20asterboylike having an external ringer or something.
18:02.32docelm0turn the volume up
18:02.42docelm0when mine rings I can hear it in the next office
18:03.27asterboyya its up to 7 max
18:03.36asterboymaybe different ring tone?
18:03.42asterboyanyone using a better tone?
18:03.43docelm0really?   Im using the default.
18:03.53asterboyya that is what I thought...its loud.
18:04.03[TK]D-FenderHey, anyone here "in the know" about SIP IM support support plans for *?
18:04.28Hmmhesaysthat would be interesting
18:04.34[TK]D-Fenderasterboy : It gets louder at 10m/s^2 after about 20s :D
18:04.37HmmhesaysI don't think it would be very hard to right
18:04.48asterboylol
18:04.52praetis the FreePBX web tool recommended?
18:04.57Qwell[]~amp
18:04.59jboti guess amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
18:05.14[TK]D-Fenderpraet : What do you WANT out of *?
18:05.19Netgeeksbacktrace: http://pastebin.ca/50449   Anyone who can read it mind taking a quick gander and tell me if you see anything that jumps out at you?
18:05.39praetI am still figuring out what its capable of :)
18:06.29[TK]D-Fenderpraet : Its capable of making you dependant on scripts noone will want to debug when you can't figure out that idiot-proof systems simply weren't designed to handle MORONS :D
18:06.30praet[TK]D-Fender: actually I would like to be able to connect from a sip phone from outside and then make calls out of the voip
18:06.57[TK]D-Fenderpraet : All easy with basic * setups
18:08.09*** join/#asterisk Op3r (n=op3r@202.71.189.70)
18:08.19Op3rdoes any one have problems with chanspy?
18:08.45*** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer)
18:08.57JerJeralright who's bitchin
18:09.03blitzrageJerJer: I am!
18:09.05Qwell[]JerJer: me
18:09.09Qwell[]fix my DID! :(
18:09.09JerJergood - keep it up
18:09.13NetgeeksI can also bitch if you need more
18:09.16Ariel_me too
18:09.21blitzrageJerJer: where's my free service?!
18:09.31blitzrage:)
18:09.34Op3rI can bitch about chanspy
18:09.41jsharpFree?  I want people to pay me to make phone calls.
18:09.44blitzrageI can bitch about bitches?
18:09.48blitzragejsharp: lol
18:09.50Qwell[]blitzrage: naturally
18:10.21JerJerwho wants to buy a VoIP provider - 800 billion in debit,  thousands of pissed off customers   :)
18:10.33JerJercheap :)
18:10.38blitzrage10 bucks!
18:10.43Dandanwho is it?
18:10.47Dandanwhat is the name?
18:10.47Dandan:D
18:10.49Op3rjsharp: how about trying to work in a call center?
18:10.52denonblitzrage: you did hear the debt part, right?
18:10.52Drukenuhmm.... no thanks, i don't need that kinda debt :)
18:10.59blitzragedenon: oooooo :)
18:11.01asterboyummm....I think you want to ask 800,000,000,010
18:11.12asterboythat way you don't incure the debt
18:11.13jsharpOp3r:  I'll pass.  I've worked at one before.
18:11.14denonJerJer: $-801B
18:11.18JerJeri'll even throw in an Asterisk sticker
18:11.24asterboylol
18:11.25Qwell[]JerJer: What, no mousepad?
18:11.29Qwell[]cheapskate
18:11.31asterboykeychain?
18:11.33Dandanno yoyo?
18:11.34asterboyT-Shirt?
18:11.36asterboyMug?
18:11.42jsharpThrow in a couple of voice prompts by Allison and you've got a deal.
18:11.50JerJeron a more serious note - shit should be fixed here soon
18:11.51Qwell[]no, stickers, mousepads, pens, and screwdrivers
18:11.54asterboyHow do you get 800 billion in debt?
18:11.55jbalcombhow do i kill a live call from the CLI?
18:11.57denonthrow in allison ..
18:11.57Qwell[]nothing else
18:12.01Qwell[]jbalcomb: soft hangup
18:12.04jsharpdenon: Well, there is that too.
18:12.08Op3rjbalcomb: soft hanguo
18:12.12Op3rerr
18:12.13JerJerasterboy:  its called a joke
18:12.16[TK]D-Fenderjbalcomb : He lives!
18:12.16Dandanjbalcomb: killall -9 asterisk or stop now
18:12.19Dandan:)
18:12.20asterboyah
18:12.34Op3rshutdown -r now ?
18:12.37asterboyHow far in debt is the US now? 8 Trillion
18:12.39jbalcombawesome!! many laughings ensue.
18:12.40Dandanthat's a bit radical
18:12.41Dandan:)
18:12.48Netgeeks800 billion in debt is easy if you are a government
18:12.48Dandanjbalcomb: you got your answer
18:12.52Dandannow we need our laughs
18:13.10DrukenJerJer: is it dead in the water or still a float?
18:13.35Dandansorry which Voip Provider are you talking about?
18:13.43DandanI am kinda out of the loop on that
18:13.53JerJerDruken:  we are very much alive
18:13.54*** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net)
18:14.00Flautohello
18:14.01Drukenexcelent
18:14.06Op3rvoip provider with .8 trillion debt? really?
18:14.14JerJerjust have to deal with yet another DID outage that is totally out of our control
18:14.19*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:14.21*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
18:14.30FuriousGeorgeanyone have any experience with adsi phones?
18:14.40Drukentelcos are fun..... REALLY.....
18:14.52paolobHi guys! Where can I find documentation about connecting to wengo with asterisk? The corresponding voip-info page doesn't work for me.
18:15.00Hmmhesaysok the manager interface is irritating me today
18:15.01Flautomy wakeup call does not hold up to the time i setup
18:15.10Flautoany idea
18:15.20asterboywhat package holds "hdparm"?
18:15.21Math`check your machine's date/time?
18:15.22Hmmhesayscheck your system time
18:15.28Math`asterboy: which distro
18:15.38asterboylfs
18:15.45Math`ah
18:15.47ManxPowerFlauto, no idea.  The standard asterisk distro does not come with a wakeup call feature
18:16.00ManxPowerFuriousGeorge, yes.  They suck.
18:16.06FuriousGeorgelol
18:16.08FuriousGeorgek
18:16.13Dandansraight to the point
18:16.18Dandan*straight
18:16.35ManxPowerFuriousGeorge, remember, ADSI use a 1200 baud FSK data stream so everything on the screen updates s l o w l y  That IS if you can get the ADSI unlock codes for the phone.
18:17.18tier_1man rewriting a dial plan is a pain
18:17.20Flautomanxpower, i installed a agi for wakeup.
18:17.23coppice120 characters/second isn't so slow for such a small screen
18:17.33tier_1the rate they keep changing things
18:17.38ManxPowerFlauto, then you should talk to the author of the AGI
18:17.43Flautowhen i schedule a call, it calls back right away,
18:17.46[TK]D-FenderManxPower : I have a P390 from when I first started with * to get rid of :)
18:17.50ManxPowercoppice, it is for ADSI phones.
18:17.57tier_1flato what wake up did you use
18:18.04HmmhesaysOk somebody tell me why my command isn't working, ARGH
18:18.10tier_1the agi or the php
18:18.11ManxPowerFlauto, How are you expecting us to help with your problem?
18:18.12Flautoiti s the one on the wiki
18:18.31tier_1there are 2 diff ones
18:18.33Flautothe php one
18:19.06ManxPowerFlauto, would you go to a GM dealer to get help with your Ford truck?
18:19.07tier_1it does not ask you what time to call you back when you call it
18:19.24Flautotier
18:19.35asterboyhow about lspci?  What package does that exist in?
18:19.42Flautoit calls me back right away
18:19.45Qwell[]asterboy: pciutils or some such
18:20.07asterboyexcellent thanks, I need to check my interrupts...getting bad echo
18:20.44*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
18:21.17*** join/#asterisk xbit` (n=xbit@frugalware.elte.hu)
18:21.20xbit`hi
18:21.21paolobGuys, how do I get asterisk registering to a voip provider when I'm behind a router without nat? Is there any specific instruction to put in the config files?
18:21.23tier_1brb
18:21.41bkw__paolob sip or iax?
18:21.55paolobbkw__, sip
18:22.05bkw__the example sip.conf has one
18:22.28bkw__its register lines
18:22.29paolobbkw__, what instruction?
18:23.03*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
18:23.06paolobbkw__, sip.conf or extensions.conf?
18:23.19*** join/#asterisk rhett (n=heyrhett@cpe-72-177-11-82.austin.res.rr.com)
18:23.29rhettdoes anyone know if you can use asterisk with skype?
18:23.34Qwell[]rhett: no
18:23.39rhettwhy not?
18:23.51Qwell[]because skype is a poor protocol
18:24.03rhettisn't there an api?
18:24.06Qwell[]no
18:24.10rhettwhat's poor about it?  it sounds good to me
18:24.13ManxPowerrhett, because Skype and Asterisk do not support any common protocols.  The Skype protocol is closed.
18:24.19asterboyThanks Qwell, downloaded pciutils from here:
18:24.20asterboyhttp://atrey.karlin.mff.cuni.cz/~mj/pciutils.shtml
18:24.31Qwell[]asterboy: why not get it from your distro?
18:24.32ManxPowerrhett, but you are welcome to try to write Skype support.
18:24.43Qwell[]in fact, if you succeed, you can claim the bounty
18:24.55rhettthere is a bounty>
18:24.57rhett?
18:24.57ManxPowerQwell, what is the bounty these days?
18:25.01asterboyso I cat /proc/interrupts and it shows my tdm card is on its own IRQ.
18:25.03Qwell[]got me
18:25.07ManxPowerARGH!  Anyone in Alabama building an ark?
18:25.15asterboyBUT, lspci -v shows that it shares with Video!
18:25.15rhettis it legal, or is skype proprietary IP and they will sue me?
18:25.18Qwell[]/nick Noah
18:25.22ManxPowerwe are now at 3 days and nights of rain.
18:25.23asterboyno wonder there is echo
18:25.25Qwell[]rhett: it's legal
18:25.29asterboyQwell, I'm on lfs
18:25.51Qwell[]rhett: neigh impossible...but legal
18:25.54ManxPowerQwell, unless they patented something, but that would only apply in countries with software patents
18:25.55coppiceManxPower: only 37 to go
18:25.55Ariel_ManxPower, wow, we are dry here and need rain. Send it to us..
18:26.15Qwell[]ManxPower: oh, they have patents, I'm sure
18:26.18ManxPowercoppice, yeah, but how many days DOES it take to build an ark?
18:26.33sivanaheh
18:26.33Strom_Cforty-two
18:26.34Qwell[]ManxPower: About 8 years...depending on size, of course
18:26.46ManxPowerQwell, I'm already behind schedule then.
18:27.06coppicepull the EHT plug of the CRT in your TV, and that can produce a good arc very quickly
18:27.17rhettdoesn't skype have the cheapest long distance?
18:27.18ManxPowerAt least the hail yesterday didn't do any significant damage yesterday.
18:27.20rhett2 cents per minute
18:27.21Qwell[]rhett: no
18:27.26ManxPowerrhett, no.
18:27.27rhettis there any cheaper way to get worldwide?
18:27.28Qwell[]rhett: That's average, at best
18:27.41rhettwho can i get cheaper worldwide with?
18:27.50Qwell[]anybody else?
18:28.10rhettI saw 20 cents per minute worldwide
18:28.10ManxPowerteliax 1.9 cents
18:28.33rhettoh neet
18:28.35rhettneat
18:28.38ManxPowerrhett, anyone that claims 2 cents/min to anywhere in the world is trying to rip you off.
18:28.47coppicewell, there the market research phase. then the proposal, with extensive power point slides, then sourcing the wood, and satisfying all the related environmental issues. then you have rare species import/export issues, vetinary quarantine, etc. its easier just to sit back and drown
18:28.58rhettHow much would it cost me roughly to set up a IVR with 50 lines?
18:29.07Qwell[]coppice: rowboat
18:29.07bkw__rhett, asterlink.com does 2.0
18:29.15*** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-11-74.w80-8.abo.wanadoo.fr)
18:29.18Qwell[]bkw__: worldwide?
18:29.22julien[re]hi there
18:29.25ManxPowercoppice, *true*
18:29.26bkw__haha NO
18:29.29coppiceI don't pay as much as 2c a minute to most of the world. there are always some places that are super expensive, though
18:29.30bkw__who ever can do that is daft
18:29.31Qwell[]bkw__: Well, that was the question :p
18:29.37julien[re]i've got a problem with a SIP phone, the volume is too low
18:29.52julien[re]can i increase the volume in *?
18:29.58rhettI need to pay $10 / month for each of the 50 lines, right?
18:30.01ManxPowerrhett, you can 2 cents /min to MUCH of the world.
18:30.16julien[re]rhett, u pay more than 2c/min to here
18:30.25asterboyQwell, good point, I checked BLFS and those utilies are there...my bad.
18:30.28ManxPowerjulien[re], Anything in a sip phone is done in the phone, not in Asterisk
18:30.49julien[re]yeah i know, but there's no way to increase the volume in *?
18:30.57ManxPowerjulien[re], what is the country code for France?
18:31.01julien[re]33
18:31.01bkw__fr
18:31.02*** join/#asterisk Thock (n=kvirc@216.119.93.253)
18:31.04julien[re]fr
18:31.05ManxPowerjulien[re], no.
18:31.10julien[re]ok cheers
18:31.10bkw__oh beat me to the 33
18:31.17julien[re]:D
18:31.32sevardTo liberia I get 29 cents a minute :\
18:31.36rhettSo, the IVR would be like $10 / month *50 lines + bandidth + server
18:31.42rhettis that it?
18:31.46Qwell[]rhett: wtf did you get those numbers?
18:31.56coppiceto Hong Kong I get zero cents a minute
18:31.57rhetti'm asking what it takes
18:32.04Qwell[]rhett: a per minute account
18:32.06rhettwhat's the cost to set up an IVR with 50 lines
18:32.07Qwell[]coppice: nice
18:32.09nahireanHeeeeeeeeeeeeee, ha helelemama Ohi'mbube!
18:32.12ManxPowerTeliax charges 2 cents a min to call france
18:32.14sevardcoppice: nice.  I'd like to put an * box in liberia and get free calling, that'd be nice.
18:32.16Qwell[]rhett: ONE MILLION DOLLARS
18:32.19ManxPowerland line, obviously
18:32.22rhettqwell, I don't have to pay per line?
18:32.22sevardTeliax charges 0.29 to liberia
18:32.25Qwell[]rhett: no
18:32.29rhettwhoa
18:32.32julien[re]i've got unlimited france ?7/mo.
18:32.39*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
18:32.48rhettfor outgoing for incoming?
18:32.49brif8where is the first place to check for random dropped calls. I process about 400 calls a day 60 and hour (at max) and will drop on average 2 - 5 calls per day. so far randomly
18:32.51ManxPowerrhett, see http://teliax.com/rates.html
18:32.51Qwell[]rhett: You need to either start reading, or hire a consultant
18:32.54Qwell[]rhett: neither
18:33.02ManxPowerteliax is not the CHEAPEST, but they have pretty good rates
18:33.19sevardManxPower: I get switching problems on their end sometimes with outbound calls
18:33.23ManxPowerbrif8, callprogress and busydetect are a common cause.
18:33.24bigblock02good luck keeping an * box up and running in liberia...lol...
18:33.27pigpenHey, what is everyone using for a -small- POE switch to power, lets say 4 polycoms?  I need a small one for demo's....
18:33.37sevardManxPower: sometimes I'll make a call and it'll ring on my end but not go through, i'll hang up and make the call again and it will go through.
18:33.37bigblock02power probably goes out a few times a day
18:33.43rhettthere are too many acronyms, i'm very new to this
18:33.57bigblock02its not like you can stuff a box in a colo or something like that
18:33.57rhettI know that VOIP takes 20-100 kbps bandwidth per line, right?
18:34.03Strom_Crhett: I'm a consultant
18:34.04julien[re]yep
18:34.06Strom_Chire me :)
18:34.08sevardhey Strom_C
18:34.11Strom_Chi
18:34.14ManxPowerrhett, perhaps you need to read The Good Book
18:34.15sevardsup bieoch
18:34.16ManxPower~thebook
18:34.18jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
18:35.06rhettThings like http://www.asteriskdocs.org do not just flat out tell you how much it costs to set up a 10 line IVR
18:35.08asterboyinteresting...* suggests that you run IDE drives in your system with a UDMA of at least 2 but no higher than mode 3.
18:35.13rhettI can't find that info anywhere
18:35.16asterboyMine is set at 5!
18:35.27asterboyanother Interrupt problem
18:35.50darkskiezreally didnt know that about the UDMA
18:35.51ManxPowerrhett, that is like asking how much it will cost to do a 10 line program.
18:36.03rhettI mean hardware
18:36.09ManxPowerFirst you have to know if a "line" is a phone line or a "line" is a IVR script line.
18:36.10brif8ManxPower: I don't see either of those in zapata.conf nor sip.conf and most calls are routed via the Internet and a VoIP provider not zap TDM44B card
18:36.11asterboyrun hdparm -i to find out what your at
18:36.16rpmwhat would someone recommend for a embedded linux system? i want to put together something quickly, basically just glibc, a shell, init and asterisk?
18:36.22ManxPowerrhett, CT1/CAS, PRI, or analog FXO?
18:36.23Qwell[]rhett: Like I said...read, or hire a consultant
18:36.30*** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
18:36.54ManxPoweror SIP or IAX?
18:36.57nahireanrpm, Fedora minimal install? .. then yum packages as you need em ;)
18:37.04julien[re]i've got another guestion:
18:37.11[TK]D-Fenderbrif8 : TDM44B ... neato... where'd you find one like that? ;)
18:37.12rhettgreat thanks for the meaningless-to-me acronyms
18:37.13rpmnahirean: i want to keep it below 50mb though.
18:37.17julien[re]does MP3Player work like Playback
18:37.18asterboywhat do you guys have for hdparm -i?
18:37.23julien[re]ie. can i get the pressed digits?
18:37.31nahireanrpm, isnt slack with just disk a's pretty low?
18:37.33ManxPowerrhett, As you can see you don't even know enough to ask a good question.
18:37.51brif8[TK]D-Fender: oops yeah one like that would be neat  I meant TDM22B
18:38.07ManxPoweryou are basically doing the same as asking "How much would it cost to build a 10MB database application"
18:38.27Qwell[]ManxPower: i want to be the next vonage, can you help me plz?
18:38.31ManxPowerThe answer is, of course, "How the hell would we know?  You've not even given us any useful information"
18:38.59nahireanQwell, I can help.  Approach VCs for roughly 500 million, then spend it all on advertising and bad CSRs.
18:38.59ManxPowerQwell, Sure!  We can start out by you /msg'ing me your banking information and your business partner's banking information.
18:39.05*** join/#asterisk uski (n=uski@ALagny-151-1-29-35.w83-114.abo.wanadoo.fr)
18:39.22*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
18:39.31[TK]D-FenderManxPower : Be nice to the n00bs!
18:39.48ManxPower[TK]D-Fender, Why?  I do NOT work for Digium. 8-)
18:40.09[TK]D-Fenderrhett : You can PM me your questions and I'll help you get started...
18:40.23uskiHi; I am veeeery new to Asterisk, and the website has mostly specialized and detailed informations for people that already know how to works. I would like to read an explanation of what Asterisk can do (I can only imagine it for now), so is there a nice guide somewhere with basic informations for n00bs ?
18:40.29coppiceits a basic human decency thing. be nice. not helpful, but nice :-)
18:40.38Cresl1ncoppice!!!!!
18:40.40Thockuski: Check out the WIKI
18:40.42Cresl1nhey :-)
18:40.49Thockuski: it has some pretty good basic info
18:40.53Qwell[]~thebook
18:40.54jboti guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
18:40.58Qwell[]uski: buy that
18:41.07ManxPowerBeing mean to stupid people is what makes life worth living.
18:41.21*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
18:41.42brif8ManxPower: where else could they be ?
18:41.43cpmflood the market with attractive ethically amiguous women dressed in business suits, with your logo on em. Instant vonage success.
18:41.44nahireanManxPower: Ignorance != Stupidity
18:41.45uskiThock, i'm not sure of which Wiki you are speaking of, could you be more specific please ?
18:41.53ManxPowernahirean, I know that.
18:41.53Qwell[]~wikis
18:41.54jbotwikis is, like, http://www.voip-info.org
18:41.58Thockuski: www.voip-info.org
18:42.04uskiok, thanks ;)
18:42.54marcus2anyone here using CAC AB-IIs for FXS ports?
18:42.57ManxPowerwhich is why I try VERY hard not to be mean to people that don't know English very well, or people that are trying to learn.  It's the people that demand we do everything for them, tutor them in basics of telephony and linux, and then set up an Asterisk server for them that I get annoyed with.
18:43.01*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
18:43.42nahireanManxPower, Couldn't it be argued that they simply aren't used to this sort of "Do-it-yourself" philosophy?  I mean, let's face it.. self sufficiency is an alien concept to most people ;) Save for us enlightened, of course. :P
18:44.04ManxPowernahirean, Yeah, but then they ignore people that tell them to go hire a consultant.
18:44.31marcus2i certainly asked a lot of stupid questions on irc when i started learning about unix
18:44.33*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
18:44.34coppiceI get some wonderful e-mails from people who's math skills seem to stop at addition, who feel I should explain the finer details of how my modems work
18:44.52ManxPowerI turn down most consulting jobs because most people don't even do the most basic project planning and requirements lists.
18:45.18*** part/#asterisk julien[re] (n=julien[r@AStDenis-103-1-11-74.w80-8.abo.wanadoo.fr)
18:45.31ManxPowerThe ones that do that stuff are a dream to work with because I can concentrate on doing techical work and not have to spend 2 weeks just trying to make the customer think about what they need and what they want.
18:45.36marcus2charging someone full tech consulting rates to do project planning and requirements gathering is a good way to make money ;)
18:45.38JerJeryou mean 1+1 doesn't equal 3 ?
18:45.40Cresl1nManxPower: yeah, that's the most annoying thing
18:45.50ManxPowermarcus2, only if you are good at it.
18:45.54Cresl1nManxPower: and then they expect their job to be ridiculously cheap too
18:45.59brif8ManxPower: Am I dropping calls because I don't have them or is there another reason ?
18:46.02FuriousGeorgeManxPower: im putting in a system for someone and they want snoms for the office and el-cheapo ananlog phones for the warehouse area.  im looking into something that will be cheap and play nice
18:46.05marcus2if you're good at both, your rate goes up a lot
18:46.07marcus2speaking from experience
18:46.07FuriousGeorgeno such animal :)
18:46.12ManxPowerCresl1n, quoting my standard rate to them usually makes them go away.
18:46.20Cresl1nManxPower: same for me :-)
18:46.38brif8what is the std. rate ?
18:46.50Cresl1nManxPower: it definitely sifts the serious people from the non-committers
18:47.02marcus2furiousgeorge; using a digium T1 card and an old used channel bank, you can get analog ports for about $30/port
18:47.25ManxPowerbrif8, MY standard rate is US$2,000/day + expenses.  Significant discounts off that rate are available for some people/companies/projects
18:47.44Cresl1nManxPower: that's pretty good
18:47.50Cresl1nI think that' s more than mine is
18:48.07Cresl1ndependent on your definition of a day
18:48.13brif8seriously $250/hr is a going rate for a VoIP consultant
18:48.16JerJerif you want your hand held, its gonna be more   :)
18:48.21Kattybrif8: oohlala
18:48.29ManxPowerCresl1n, my "standard rate" could also be called me "I don't want to do this project, but if you give me enough money I'll do almost anything rate"
18:48.46Cresl1nManxPower: yeah, I certainly know how that goes
18:48.58ManxPowerJerJer, I usually make people buy me dinner and drinks before I'll hold their hand.
18:49.16brif8all jokes asides what is a true acceptible rate for a VoIP consultant ?
18:49.26Cresl1nbrif8: that's about right
18:49.38FuriousGeorgemarcus2: they currently have * sitting under their old pbx with a 2 tdms and existing analog ports
18:49.41marcus2anywhere from 75-300/hr
18:49.48Cresl1nbrif8: for somebody that knows what they're talking about, I would say that's fairly expected
18:49.51FuriousGeorgethey want to use their existing ports for some of the phones
18:49.57FuriousGeorgewhich i cant blamem for
18:50.00*** join/#asterisk codebreaker (n=codebrea@xserver.flexserv.de)
18:50.03brif8Cresl1n: is that area specific or general across the entire US
18:50.14Cresl1nbrif8: it's per person specific
18:50.16Cresl1n:-)
18:50.22brif8I C
18:50.31marcus2furiousgeorge; if they already have the ports, whats the problem? ;)
18:50.47FuriousGeorgemarcus2: one line analog phones either suck or try to do to much
18:50.56marcus2jerjer; so you guys already have someone to move the 800 DIDs to?
18:51.00brif8for a minute I thought the company was getting a good deal, they just had one in charging $85/hr
18:51.00FuriousGeorgethe only problem is shopping for them
18:51.11marcus2furiousgeorge; oh, heh.  go to frys ;)
18:51.11FuriousGeorgehence my question earlier about adsi
18:51.15Qwell[]brif8: That's quite cheap
18:51.23marcus2oh you want some intelligence in the cheapo phones?
18:51.37ManxPowerYou can easily get $300/hr in New York City, but you'd have trouble getting even $60/hr in New Orleans
18:51.57[TK]D-Fenderbrif8 : My rates are far more affordable :)
18:52.07brif8Qwell[]: yeah from what is just said. In this area the highest IT salary is probably $45/hr my boss thought the guy was hussling.
18:52.08coppiceeven if you bring your own submarine?
18:52.15marcus2i dunno about that, i know someone who went to new orleans to do volunteer recovery work and was offered 175/hr to stay around doing telephony stuff
18:52.16Cresl1nheh
18:53.02ManxPowerbrif8, An hourly consultant is not an employee.
18:53.05codebreakerif i call from iax-client > pstn i normaly should hear somthing like tuuut tuut but i hear krangbangterriblebang(at the iax-clint)  after i take the line on pstn-clint i have wonderful connection. only the "tuut tuut" is teribble any hint?
18:54.04ManxPowerIf you factor in health insurance, unemployment insurance, vacation, workman's comp, SSI, heating, cooling, power, phone, and simple rent of the office the person uses, etc, you are paying MUCH more than $45/yr for top IT salary.
18:54.05blitzrageis there a way to tell Dial to stop trying to dial an endpoint if the other side doesn't answer after X number of seconds?
18:54.08brif8ManxPower: yeah we have a network consultant  (planner & installer) she (yes that is right) charges $45/hr and for this area (backwater FL) that is high
18:54.09bigblock02(JerJer): Who is the new TF provider going to be? When will they be up and running? I want to know if I should stop the port away from nufone or not.
18:54.15ManxPower$45/yr == $45/hr
18:54.45*** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net)
18:55.13JerJerbigblock02:  you always have the right to port your number away
18:55.22marcus2jerjer; yes, but should we?
18:55.25marcus2or are you dealing with it
18:55.29brif8now we're saying the VoIP guy charging $75/hr is cheap. not in this area
18:55.30marcus2i guess i wasnt that clear from the last email
18:55.37MikeJ[Laptop]if $45/yr == $45/hr for you I have a few bucks inmy pocket :P
18:55.39FuriousGeorgesays here this phone has a "data port"
18:55.42bigblock02i dont want to be down forever and business depends on TF service!
18:55.48marcus2i certainly dont care if its down for a couple of days, i just want to make sure it comes back eventually
18:55.57JerJerwhat was not clear about back up in a few hours or so ?
18:55.58FuriousGeorgewhat kind of data port would an att analog one line phone have
18:56.07ManxPowerFuriousGeorge, an analog port.
18:56.18ManxPowerya know for those new fangled thingies called "modems"
18:56.20ManxPower8-)
18:56.31FuriousGeorgeha
18:57.17b00mer_anybody using chan_sccp for a production phone system?
18:58.16brif8reading "thebook" how would one split * over mupltiple servers ?
18:58.20b00mer_before I go buy a few cisco sccp only phones to test, I wanted to know if its ready for a production asterisk system
18:58.38marcus2jerjer; i guess its clearer, now that i'm reading it when i'm more awake
18:58.39b00mer_brif8 : what's the book?
18:58.40bigblock02well as per the email sent out... it appears like service will be down for more than a few hours!
18:58.51brif8~thebook
18:58.52jbotit has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
18:58.53bigblock02unless you can get all of the DID's ported in a few hours...
18:58.55Qwell[]b00mer_: No, because any time asterisk changes, you'll have to wait for SCOgio to update chan_sccp
18:59.03Qwell[]JerJer: ^ :p
18:59.11bigblock02so i was curios who that was so i can decide if i can stop my port request
18:59.41b00mer_ahhh that book... I've got that book... cool
18:59.48brif8b00mer_ testing two 7920 using chan_sccp right now
19:00.02b00mer_Qwell[]: is there chan_sccp2 same feelings?
19:00.03Qwell[]brif8: Did they work on chan_skinny?
19:00.04brif8all other cisco phones use SIP
19:00.10Qwell[]b00mer_: There is only one chan_sccp
19:00.17bigblock02(JerJer): I'm just worried that a couple hours will leak into the weekend and into next week
19:00.23Qwell[]the others have died
19:00.25bigblock02and come monday i'm going to have some serious problems.
19:00.38brif8Qwell[]: didn't try chan_skinny and I'm still trying to prove > 4 lines (you mentioned 6 yesterday)
19:01.38JerJerbigblock02:  If Telesthetic would have let this process complete normally there would be no outage right now
19:01.52*** join/#asterisk stack_ (n=stack@63.239.190.202)
19:02.12JerJereverything is totally out of my hands right this second
19:02.13nahireanwhoa.. there's the buck, it just got passed. :P
19:02.25bigblock02im not concerned about whats going on with telesthetic... TF service doesnt work!
19:02.36JerJertelethetis provided us TF
19:02.45bigblock02and you provide for me...
19:02.50docelm0JER!
19:02.53JerJerthen we elected to move providers
19:02.59*** join/#asterisk boch (n=fran@unirc.com.ar)
19:03.01bigblock02who and how long?
19:03.03JerJerso they cut us off
19:03.18brif8Qwell[]: how far have you tested chan_skinny ?
19:03.21stack_Is there anything special that needs to be done with Asterisk to support Same Line Disconnect (or whatever you want to call it)?
19:03.32Qwell[]brif8: not very, heh
19:03.39ManxPowerstack_, I am not familiar with that term.
19:04.23*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
19:04.27stack_ManxPower: If I get an inbound call and then route it to an external line, the line is broken from our switch, so we aren't using two lines for it
19:04.32Qwell[]ManxPower: home depot?
19:04.33bigblock02hello?
19:04.49brif8Qwell[]: are you working on it or what is the current status ?
19:04.52*** part/#asterisk pjz (n=pj@66.219.59.183)
19:04.59LostFrogstack_: Native transfer?
19:05.01Qwell[]brif8: yeah, when I have time.
19:05.08brif8i c
19:05.11*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
19:05.15filebigblock02: 1. This isn't the NuFone support channel, take it privately or to #nufone and 2. JerJer is doing the best he can with this situation and right now it's out of his hands
19:05.15ManxPowerstack_, "conference drop and transfer" is what it's called in the USA.  Not easy to do with analog lines in Asterisk, impossible to do with PRI.
19:05.18ManxPowerQwell, I said CHEAP.
19:05.19stack_LostFrog: yep I believe that is another way of saying it... there are like 10 different terms for it
19:05.21Qwell[]I'm not being paid to write it or anything
19:05.32bochis it possible to connect 48 sipura spa2100 behind the same nat ?
19:05.38Qwell[]boch: sure
19:05.48jsharpIf the nat is smart enough, yes.
19:05.55bochgreat
19:06.02CoffeeIV_<PROTECTED>
19:06.07ManxPowerboch, you should be able to do it for any NAT router that was not designed by a moron.
19:06.20Qwell[]ManxPower: read: dlink
19:06.22LostFrogThere.. a smart NAT.
19:06.28Qwell[]or, rather, not dlink
19:06.28*** join/#asterisk loud (n=ariel@cypher.punk.net)
19:06.34bochlinux plus iptables?
19:06.39stack_ManxPower: we have PRI and some analog lines, if we got an incoming call on the PRI and redirected to an 800 number via the analog lines, would I be able to do "conference drop and transfer"
19:06.56Qwell[]boch: sure, if it wasn't designed by a moron
19:07.04ManxPowerboch, remember that if user A calles user B the audio still goes thru Asterisk
19:07.10stack_CoffeeIV_: I might be, there are a tons of different ways to say it, noone uses the same term
19:07.13tainted-anyone tried voip over EVDO or some kind mobile data service?
19:07.24Qwell[]tainted-: The latency is too much
19:07.38tainted-really? i've heard that EVDO is under 100 ms
19:07.40bochManxPower that's right :S
19:07.46terrapenanyone made a T1 crossover cable?  will they work going from port-to-port on Digium's PRI cards
19:07.48bigblock02(file): 1. oh I forgot this is the nufone sales channel
19:07.51ManxPowerstack_, no (unless you have a very specific telco switch with a very specific protocol setting).  See the Wiki for more information or the mailing list.  you now know everything I know about the subject.
19:08.11coppiceterrapen: crossover cables work
19:08.20CoffeeIV_stack_: I think it's also called TBCT.  Digium support told me it was supported in recent versions of asterisk as long as you had a "DS100" switch, and I'm still trying to figure out if I do or not
19:08.31ManxPowertainted-, no.  If I can't SSH over a link without hurting every time I type something, I won't even try VoIP 8-)
19:08.38terrapenit may be that this redfone fonebridge is just hosed up or configured wrong
19:08.43ManxPowerCoffeeIV_, DSM 100
19:08.47terrapeni'm not getting link
19:08.47*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
19:08.50filebigblock02: JerJer rarely sticks around here, and most of the "sales" is done by word of mouth from people in here
19:08.54loudstill with satellite manx ?
19:08.58tainted-ManxPower which service do you use?
19:09.06ManxPowerloud, Yes.  *sob*
19:09.15loudwhich sat modem do you have ?
19:09.16CoffeeIV_ManxPower: right, DSM 100, thx
19:09.22bigblock02well he was obviously was here and when asked the hard question he dodged and went to get coffee or something
19:09.22ManxPowertainted-, DirecWay and Verizon
19:09.55LostFrogFrom what JerJer has said today, I wouldn't think it was a sales pitch.
19:09.55ManxPowerVerizon "BroadbandAccess" service is not TOO bad, but you have to be in a city where it is supported.
19:10.01ManxPowerloud, DW6000
19:10.03filebigblock02: ok, so deal with it - the fact is your TF is down, he's trying to get it ported to a new carrier, so give him time
19:10.18Qwell[]bigblock02: He answered your question several times.  If you don't like it, he said that you should port it to another provider
19:10.34ManxPowerloud, once the property is wired then I'll get a voice/data T-1 and sell service to the people here.
19:10.35fileand if you need carrier grade service, then get a PRI with an SLA
19:10.37Qwell[]So, stop trolling, and take the discussion to the appropriate place
19:10.42bigblock02with the response ive been getting, sounds like a good idea
19:11.02terrapenPRI will make you so much happier anyway
19:11.08ManxPowerbigblock02, I would never use VoIPoInternet for a primary business service.
19:11.16terrapeni couldn't agree more.
19:11.17bigblock02actually, a better idea is to start issuing a chargeback for services not rendered
19:11.26*** join/#asterisk Maxxed (n=whyman@65.59.245.122)
19:11.27loudthat depends on your network.
19:11.43Qwell[]services not rendered?  outgoing calls still work. :)
19:11.57terrapenI have very nice network connectivity and I've never been happy with VoIPoInternet
19:12.11bigblock02i pay for TF termination
19:12.13filethe terms and conditions of use specific state about this
19:12.22ManxPowerbigblock02, JerJer is doing everythnig he can.  If he knew when the service will be back he would have said something.
19:12.24Maxxeddoes chanspy work in 1.2.7.1 ?
19:12.27[TK]D-FenderThe ITSP I use at home ROCKS :)
19:12.35Maxxedi see the module is loaded n all
19:12.41dlynes[TK]D-Fender: Thanks for the suggestions about the drivers yesterday
19:12.42bigblock02since when does a regulated CLEC cut people off?
19:12.43LostFrogMaxxed: It works in 1.2.7
19:12.43Maxxedbut i cant get it to work for me ;\
19:12.45Maxxed-- Executing ChanSpy("SIP/2666-a4c6", "scan|q") in new stack
19:12.46terrapenI'd use it for home.  but not for a biz
19:12.47Maxxedand thats it
19:12.47dlynes[TK]D-Fender: That was the problem, exactly
19:12.55nahireanManxPower, Have you ever heard of a carrier just "Pulling the plug" for no reason?  I mean, if they were getting calls - they should've been enjoying the cash from it, no?
19:12.56bigblock02or didnt they get paid?
19:13.00Maxxednit 1.2.7.1 ?
19:13.06Qwell[]bigblock02: Go elsewhere, you're trolling now
19:13.09Maxxeds/nit/not
19:13.26fileeveryone drop the subject now, and take it elsewhere - that includes regulars
19:13.31terrapenheh
19:13.34Qwell[]file: but, but!
19:13.38Qwell[];)
19:13.38[TK]D-Fenderdlynes : What was I helping you with again? :)
19:13.43terrapenmust....make.....redfone fonebridge....work
19:13.45ManxPowernahirean, happens all the time in the VoIP world.
19:13.47Maxxedhi'a file :)
19:13.48dlynes[TK]D-Fender: The moh problem :)
19:14.20Ariel_terrapen, have you called them for support
19:14.24dlynes[TK]D-Fender: Normally, I only have absolute necessary drivers loaded, but for some reason on that system, I had a bunch of crap loaded
19:14.26Maxxedlostfrog: so, chanspy dont work in 1.2.7.1 ? but it does in 1.2.7 ?
19:14.32terrapenariel, yep, and they're helping me
19:14.38bigblock02i wouldnt be here if nufone's customer service number was working... well i guess i'll email and look for jerJer's cell phone number
19:14.41terrapeni'm just waiting for a response back on the stuff i've sent them
19:14.42Ariel_who mark or brett
19:14.42LostFrogI don't know about 1.2.7.1, I am only running 1.2.7
19:14.45terrapenmark
19:14.46wunderkinMaxxed: you are specifying "scan" for the channel
19:14.53Maxxedwuner: yep
19:14.57Maxxedbleh
19:15.05Maxxedwunderkin: yes
19:15.20Maxxedexten => 1337,1,ChanSpy(scan|q)
19:15.29terrapenariel, I can't get a link light on the PRI ports and zttool(8) shows state "RED"
19:15.41Ariel_bigblock02, go to there freenode location #nufone
19:15.44wunderkinMaxxed, show application chanspy
19:15.48Maxxed# and * dont seem to do anything as well
19:15.55Maxxedwunderkin: its there
19:15.56terrapenAriel, are you using one of these, too?
19:16.22ManxPowerterrapen, use a T-1 crossover cable.
19:16.23Ariel_terrapen, not here I have a customer that has 3 of them and I know them all at RedFone
19:16.52Maxxedi've recompiled and checked the md5sum on the module aginst what i have now
19:16.54Maxxedits the same
19:16.58Maxxedim a loss
19:17.01terrapenariel, any chance that you could send me your configs?
19:17.05ManxPowerMaxxed, are you sure it's not a DTMF provlem?
19:17.07terrapen(zaptel.conf, redfone.conf)
19:17.20MaxxedManxPower: zapbarge works fine
19:17.21Ariel_terrapen, no there my customers
19:17.25terrapenariel, also, have you ever tried looping it back to itself?
19:17.54ManxPowerterrapen, USUALLY you use a stright thru T-1 cable to go Telco<->Asterisk and a CROSSOVER T-1 (NOT Crossover Ethernet) cable to go Asterisk<->channel banks
19:17.54terrapenariel, oh, i thought you might have set them up and maintain them or something...
19:18.19terrapenmanx, I'm going crossover T1 from PRI port to PRI port
19:18.20Qwell[]ManxPower: crossover to channelbanks?  hmm
19:18.32wunderkinMaxxed: the first parameter specifies the BEGINNING of the channel name to monitor, again: show application chanspy; you are specifying channels to spy beginning with scan
19:18.42terrapenI'm plugging one of those fonebridge's PRI ports back to itself but I get no link light
19:18.54terrapenI'm thinking that the fonebridge is not initializing itself correctly
19:19.26Ariel_terrapen, have you tried a normal cable
19:19.30Maxxedwunderkin: um, mind shooting me an example? i dont quiet follow
19:19.45*** join/#asterisk backblue (n=moo@87-196-77-122.net.novis.pt)
19:19.46terrapennormal T1 cable?  nope, not yet
19:19.50Maxxedwunderkin: what i have now.. exten => 1337,1,ChanSpy(scan|q)
19:19.54terrapenI suppose I could make one.
19:20.01Maxxedwunderkin: what am i missing
19:20.06*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
19:20.08terrapenafter we close tonight, I'm gonna try plugging the real PRI in
19:20.10wunderkinMaxxed: Agent/ to scan agent channels, SIP/ to scan SIP channels, etc, or nothing
19:20.36Ariel_terrapen, you can try a loop back cable for testing
19:20.54Maxxedwunderkin: ChanSpy(SIP/scan|q) ?
19:21.07terrapenahhh, that's a good idea, too
19:21.11Qwell[]Just SIP/, I think
19:21.21terrapeni need to make one of those, anyway
19:21.23Maxxedw/o scan ?
19:21.31Qwell[]where did you get the word "scan"?
19:21.39Maxxedvoip wiki
19:21.47[TK]D-Fenderdlynes : Oh yeah... you're welcome :)  I help so many people here I can't keep track of what I've done for whom....
19:21.50Maxxedim gona trial and error a few things ;)
19:21.52Hmmhesayswow the manager is just wonderful today
19:21.53Maxxedbrb
19:22.30*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
19:22.50Maxxedah!
19:22.51Maxxedok..
19:22.55Maxxedman do i feel silly
19:23.03Maxxedyeah, SIP|q worked fine
19:23.06Maxxedheh..geesh..
19:23.09Maxxedthanks you guys ;)
19:23.45dlynes[TK]D-Fender: Well, I just thought I'd let you know so that if you encounter someone else iwth the same problem you know how to help them
19:25.00ManxPowerMaxxed, NEVER look at the Wiki if you can get the info from Asterisk's CLI or Asterisk's docs that are included in the source tree.
19:25.06dlynes[TK]D-Fender: But, if the stupid zaptel drivers didn't load regardless of whether you have the hardware or not, it wouldn't be an issue
19:25.21*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
19:25.33*** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz)
19:25.34dlynes[TK]D-Fender: Most normal drivers will refuse to load if the hardware isn't rpesent
19:27.07tainted-Qwell do u have mobile broadband?
19:28.30LostFrogThanks people, I never new the argument to ChanSpy was a prefix.
19:29.16Strom_Cdlynes: there's a use for zaptel if the card isn't present
19:29.22Strom_Cit's called "ztdummy"
19:30.13[TK]D-Fenderdlynes : Hey.. what do I know!~ I absorb and spew.. thats all.
19:30.34dlynesStrom_C: I'm not talking about the zaptel.ko module though
19:30.50dlynesStrom_C: I mean like tor2, wcfxo, ...
19:31.26blaylockhas anyone here configured asterisk to use an analog line as a backup to the internet for outgoing calls?
19:31.30FuriousGeorgei gotta interface chan/dsp with a "horn" these people got setup.  it looks like a giant megaphone mounted on the roof, and it hooks into their old pbx via what looks like a plain old analog jack...  it must have its own power being fed into it from somewhere cuz its way loud
19:31.43dlynesblaylock: yes
19:31.47Strom_Cblaylock: yes
19:31.57FuriousGeorgecorrection:  it interfaces with a station on their old pbx
19:31.57dlynesblaylock: probably almost everyone, actually :)
19:32.10terrapensweeeeet!
19:32.15terrapenthe redfone guys fixed it
19:32.22Strom_CFuriousGeorge: it's probably connected to an amplifier
19:32.24terrapeni had a config file typo, doh
19:32.25*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
19:32.42blaylockcool
19:32.59FuriousGeorgeStrom_C: if it is, i dont see the thing.  like i said it looks like a megaphon, so i think it is its own amp
19:33.12Ariel_terrapen, great to hear it.
19:33.26Ariel_terrapen, sometimes typo's are the hardest thing to find.
19:33.27dlynesblaylock: I don't mind people messaging me, but ask first
19:33.27timscottFurious, maybe it's built into the horn.
19:33.29brodiemI have a long flight coming up.. anyone recommend a book over TFOT?
19:33.39blaylockdlynes, oh sorry
19:33.44Ariel_War and pease
19:33.54Strom_CFuriousGeorge: it has a mini phono plug right?
19:33.54jsharpWar and peas
19:34.01FuriousGeorgeStrom_C:
19:34.02*** join/#asterisk miller7 (n=none@adsl19-22dynamic.athens.acn.gr)
19:34.02FuriousGeorgeyes
19:34.02LostFrogCurious George.
19:34.05blaylockdlynes, im used to just doing it without asking
19:34.10dlynesblaylock: But, the stuff you're looking for can be found by checking dialstatus to determine whether the call succeeded or not
19:34.26timscottbrodiem: winston churchill's books about WWII are quite good, and can be read in about 30-40 hours.
19:34.31Strom_CFuriousGeorge: so, easy.  set up an extension to dial chan/dsp
19:34.35miller7anyone here using Gentoo? I can't seem to make /dev/zap/pseudo on my 2.6 kernel, so I can't test meetme
19:34.37Strom_Cone line of dialplan
19:34.38blaylockdlynes, well i think my situation is kinda unique really
19:34.40Strom_Cdone
19:34.43dlynesblaylock: yeah...unless it's something that requires a lot of texting, I prefer to do it in the channel
19:34.51*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
19:34.51terrapenariel, well, I'm not out of the woods yet
19:34.57blaylockdlynes, we will be talking to our main asterisk PBX from a customer site
19:34.57dlynesblaylock: that way everyone else that has the same problem can learn from it, too
19:34.59terrapennow the lights on the 4-port PRI card are blinking red
19:35.08blaylockdlynes, ok
19:35.20dlynesblaylock: and your situation isn't unique
19:35.21terrapenon for 4 secs, off for 1/2 sec
19:35.27ManxPowermiller7, What Zaptel timing device are you using?
19:35.29afrosheenattn polycom users, who has the latest sip and firmware for the ip5xx phones?
19:35.38miller7ManxPower: none
19:35.38dlynesblaylock: I would imagine anyone that wants a stable phone system needs that feature
19:35.39blaylockdlynes, so we will use IAX at the customer site, and then if their net fails, use the analog line to make calls instead
19:35.41terrapenafro: where did you buy the phones
19:35.43brodiemtimscott, lol how about something voip/telephony related
19:35.45ManxPowermiller7, then you can't use meetme.
19:35.47blaylockdlynes, yeah as do I
19:35.51afrosheenterrapen: a variety of sources
19:35.54blaylockdlynes, ive just never done it
19:36.00miller7ManxPower: sigh... I must use zaprtc, right?
19:36.00timscottbrodiem, oh. :S
19:36.04terrapenafro, did you buy any from voipsupply?
19:36.12ManxPowermiller7, that would be a zaptel timing device.
19:36.14afrosheenterrapen: yeah a few
19:36.15dlynesblaylock: anyways...what you need to do is after the Dial application completes, do some call handling based on the dial status
19:36.20ManxPowerbut ztdummy would be preferred on 2.6
19:36.30ManxPowerI don't even know if zaprtc workd on 2.6
19:36.32dlynesblaylock: You need to determine if it was congested, busy, answered, ..., and branch appropriately
19:36.34blaylockdlynes, right but how do i set this up?
19:36.41dlynesblaylock: in your dial plan
19:36.46terrapenafro, email your voipsupply rep and ask for the firmware.  they will send you a login for their ftp server, which has them all
19:37.07blaylockdlynes, so this can all be done from the dial plan?
19:37.18dlynesblaylock: correct
19:37.20afrosheenterrapen: thanks buddy, I didn't think they would provide that, thenerds.net and another company didn't
19:37.24ManxPowerblaylock and dlynes and exmaple of post Dial processing is included in extensions.conf.sample  See [macro-srd-exten]
19:37.38dlynesManxPower: I don't need help with it...I'm already doing it :)
19:38.07terrapenafro, they will be glad to
19:38.13blaylockManxPower, cool thanks
19:38.15afrosheenterrapen: ok cool
19:38.26dlynesblaylock: but yeah it's actually [macro-stdexten]
19:38.35dlynesblaylock: i think that's where I copied it from, actually :)
19:38.39ManxPowergreat.  here comes the thunder.
19:39.10blaylockdlynes and ManxPower, thanks a bunch
19:39.32blaylockthat was much faster than having to read a bunch of docs and figure it out
19:39.36blaylock:-D
19:39.47[hC]Hmm, if im showing 100% volume in ztmonitor on one Zap channel (FXO port in a tdm400p) and listening in its just full blown static, which is most likely the problem, the fxo port or the line?
19:39.53LostFrogI'm afraid when chanspying that people can here me.
19:40.04dlynesblaylock: Yeah, but if you read the dialplan docs, you can find out what other dial status variables aren't accounted for in macro-stdexten
19:40.06ManxPower[hC], IRQ conflict?
19:40.22[hC]ManxPower: even though the other two fxo ports installed are fine?
19:40.25ManxPower[hC], It is a direct POTS line?  Not connected to a PBX?
19:40.40[hC]ManxPower: 3 POTS line into 3 individual FXO modules on a TDM400P
19:40.43*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
19:40.49ManxPower[hC], it's brain dead easy to check for IRQ conflicts, even my can could do it.
19:40.56ManxPowercan == cat
19:41.15ManxPoweror better yet: Even my cat /proc/interrupts could do it.
19:41.17[hC]Maxxed: its sharing an interrupt with eth0
19:41.19timscottthat's one smart can.
19:41.21[hC]er
19:41.22[hC]manx.
19:41.32[hC]ManxPower: IRQ169 has wctdm and eth0 on it
19:41.34ManxPower[hC], well, there is your problem.
19:41.35*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
19:41.45*** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net)
19:41.47[hC]ManxPower: However, Would an IRQ conflict not affect the entire card, not just one module?
19:42.05asterboyin lilo or grub, is there a way to add a boot option like vga=normal and acpi=fuckoff??
19:42.10*** join/#asterisk swytch (n=ezcall@d83-177-197-138.cust.tele2.fr)
19:42.21timscottasterboy, yes.
19:42.23ManxPower[hC], IRQ conflicts can make frogs rain from the sky, destroy entire cities, or make just some of the modules on a card not work.
19:42.26*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
19:42.41Ariel_it's magic
19:42.43jsharpIRQ conflicts are the true cause of global warming.
19:42.43[hC]Hah
19:42.59Qwell[]jsharp: I think you mean IRC conflicts
19:43.03asterboymine bypasses the boot:, what are the parameters to add to say lilo.conf?
19:43.11Qwell[]asterboy: noacpi?
19:43.13Ariel_there is no global warming... just history repeating it's self
19:43.26Hmmhesaysis there any way I can get show channels to display the entire channel name?
19:43.29tasatquestion about play tones in an agi:  does it need to be followed by a wait?  everything I read says the tone should continue until 'stopplaytones' is called. Any clues?
19:43.34[hC]Okay, I figured it would have done the whole card.. Anyways then, this may sound stupid, but if i dont have console access to that machine, and cant go into the bios to move irq's around (I dont htink i can anyways, its a shitty dell) how do i force things to different IRQ's from at least grub?
19:43.35timscotti know with grub it's `vga=whatever` and `noacpi`
19:43.37Hmmhesaysright now it cuts off part of it
19:43.38afrosheenasterboy: noapic nolapic is fun also
19:43.39[hC]I guess I could try acpi off
19:43.52asterboyjust add those anywhere in the boot configs?
19:43.57[hC]timscott: acpi is acpi=off, APIC is noapic
19:44.01Ariel_Hmmhesays, sip show cahnnels
19:44.02*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
19:44.02Qwell[]asterboy: at the end of your kernel line
19:44.07ManxPower[hC], you can't.  you need to physically move the card around
19:44.08[hC]and i guess I meant noapic :)
19:44.11timscott>_<
19:44.15afrosheen:)
19:44.19timscottmy bad
19:44.27DelmarAre there known issues compiling zaptel 1.2.5 ? cuz mine is blowing all over the place.
19:44.28[hC]ManxPower: this must have happened after a recent reboot, cause it was fine previously, before a recent reboot.
19:44.46Ariel_Delmar, CentOS
19:44.53Ariel_~Cenosbug
19:45.01Ariel_~Centosbug
19:45.02jbothmm... centosbug is a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package.
19:45.02[hC]I'll dick with it later.
19:45.09Qwell[]Ariel_: You lose
19:45.11DelmarAriel_, explain?
19:45.28Delmarok
19:45.40Delmarim running 2.6.14
19:45.54Ariel_Qwell[], what?
19:45.57Qwell[]Ariel_: nothing
19:45.59Qwell[]bot hates you
19:46.04Ariel_typo
19:46.20asterboyok, thanks guys!
19:46.44dlynesWhy would a distro intentionally break the kernel source?
19:46.51HmmhesaysAriel: sip show channels doesn't give the channel name
19:46.52timscotte-tards
19:46.58Hmmhesaysshow channels, cuts off part of the channel name
19:47.24Ariel_Hmmhesays, show channel name-tab
19:47.43afrosheenwell ask Redhat why they broke legacy scsi support which affects VMWare :)
19:47.55afrosheenRH has jacked up their last 2 kernels
19:48.03dlynesWhy not just leave the damned kernel alone?  It works just fine
19:48.04Ariel_dlynes, type
19:48.06swytchmaybe because vmware is commersion (too)
19:48.06Ariel_typo
19:48.23swytchcommersial
19:48.38dlynestypo
19:48.40Hmmhesayscan't do that from the manager interface
19:48.41DelmarAriel, regarding that jbot message... im pretty sure it doesnt apply to me. firstly im not running Centos.. nor am i running kernel 2.6.9, further more the file it refers to has 230lines ... so there is no line 407 ::P
19:48.41dlynesAriel_: and?
19:48.58Hmmhesaysi'm trying to get the channel names of two channels so I can redirect them
19:49.23Ariel_Delmar, ok that is your error's use pastebin.ca
19:49.28dlynesDelmar: What's the error message you're getting?
19:49.37*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
19:49.38*** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com)
19:50.15DelmarIm running Debian SID, with kernel 2.6.14.. when i try to compile zaptel 1.2.5 .. there is tonnes of error output ending up with make: *** [linux26] Error 2
19:50.20miller7ManxPower: Thanks, I modprobed crc-ccitt and now everything works fine with meetme. Thanks again
19:50.35dlynesDelmar: Are you running 'make linux26', or just 'make'?
19:51.12Delmardlynes, actually I have tried both.. and i do have a symlink  of linux-2.6 pointing to /usr/src/linux-2.6.14
19:51.37*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
19:51.38dlynesDelmar: Try doing ln -s /usr/src/linux-2.6/ /usr/src/linux as well
19:52.07dlynesDelmar: It's bloody stupid, but they're using hardcoded directory names in the makefile
19:52.36Delmarnone of that makes any diff
19:52.37dlynesDelmar: Instead of reading it from the /lib/modules/`uname -a`/build symbolic link
19:52.39mogormanir osss....
19:52.45mogormanerr its oss make a patch....
19:53.08dlynesDelmar: pastebin your make output?
19:54.20dlynesbrb
19:54.23Delmarim just going to try and compile some older zaptel... if they fail.. I will know its my system and can start thinking about that
19:54.30LostFrogme curses at audiocodes for not having GSM on their MP-1XXs
19:54.39dlynesDelmar: zaptel 1.2.5 works just fine, though
19:54.49dlynesDelmar: I've compiled with no problems with 2.6.15.5
19:55.16Delmaryeah something must be wrong with my system. bugger. thats all i need to hold me up today.. i have lots to do. fuckin linux. grrr
19:55.20dlynesDelmar: That's with a custom compiled kernel though
19:55.34dlynesDelmar: Are you getting a whole bunch of could not find this symbol or that symbol kind of errors?
19:55.48*** join/#asterisk MikeJ__ (n=vircuser@64.241.37.140)
19:56.20*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
19:56.49dlynesbbiam
19:57.20tasatTrying again... I'm having trouble getting playtones to play tones in the background without a wait when executed by an agi. Is this possible?
19:57.40luke-jr_Are there any conference applications that don't need zaptel drivers?
19:57.48stack_Is there a way to do distinctive ring (or something applicable) with the polycom 301/501?
19:58.22tainted-luke-jr_ meetme
19:58.34luke-jr_tainted-: that requires zaptel, last I checked...
19:58.49brif8how well does ChanSpy work in a production env.
19:59.05tainted-luke-jr_ really? i don't have zaptel installed and it works fine for me
19:59.23luke-jr_tainted-: interesting...
19:59.29*** join/#asterisk dlynes (i=1000@S010600c09f9a0fc4.vc.shawcable.net)
19:59.32tainted-luke-jr_ even if it were required, why not just install it
19:59.33Qwell[]tainted-: it won't even compile unless zaptel is installed
19:59.45Delmardlynes, gotta be something dumb like headers out of synch.
19:59.50Delmardlynes, hopefully its that simple
19:59.53tainted-really
19:59.53Delmarmust be
19:59.59bigblock02no response in #nufone about this issue... looks like the @'s were just trying to get me out a publicly viewed channel so nufone can blow me off
20:00.15*** join/#asterisk boddy (n=boody@85.102.209.112)
20:00.16tainted-bigblock02 lol
20:00.16dlynesDelmar: After you created that /usr/src/linux symbolic link
20:00.21tainted-another happy customer?
20:00.23*** join/#asterisk iCEBrkr (n=icebrkr@69.9.167.70)
20:00.28Qwell[]file[laptop]: do it!
20:00.30dlynesDelmar: Did you do a make clean ; make, or just make?
20:00.44luke-jr_tainted-: I don't just install random drivers
20:00.50Delmarsure i always make clean when compiling stuff
20:00.52luke-jr_tainted-: what * ver?
20:00.55file[laptop]Qwell[]: meh
20:00.55dlynestainted-: another troll :)
20:00.56Delmarill just see about these headers.
20:01.25nahireanmay I ask how he's trolling?  he just wants his service to be back up.. surely we can all relate to that hehehe :)
20:01.28*** join/#asterisk apardo (n=apardo@87.217.146.193)
20:01.36tainted-dlynes w/ nufone it' justified.. but this channel won't offer sympathy
20:01.37blaylockdlynes, true, probably still read anyway
20:01.42*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
20:01.45Qwell[]nahirean: There is a right place, and a wrong place.  This is the wrong place
20:01.49dlynesblaylock: pardon?
20:02.01bigblock02i tried #nufone for the last 20 minutes
20:02.04bigblock02no one peeps
20:02.07ManxPowerbigblock02, What do you expect us to do about your problem?
20:02.12ManxPowerIf you want a therapist, go find one.
20:02.16blaylockdlynes, oh sorry...you were talking about the docs for that failover thing a while ago
20:02.22bigblock02fix it?
20:02.23Qwell[]bigblock02: Drop it, now
20:02.35dlynesbigblock02: How are we going to fix it?  We don't work for nufone
20:02.41tainted-i think he is just trying to make it known that nufone is doing him wrong publically
20:02.49nahireanI agree tainted
20:02.52Qwell[]tainted-: Jeremy made it public himself! :P
20:02.58*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
20:02.59ManxPowerbigblock02, I see you have a rich fantasy life.
20:03.09bigblock02you see a lot, you know nothing about me!
20:03.15file[laptop]drop it NOW
20:03.18tainted-bigblock02 just port out of nufone dude
20:03.28ManxPowerbigblock02, Correct, I see only what you present on the channel.
20:03.36tainted-bigblock02 it's not worth it
20:03.41Qwell[]:D
20:03.46file[laptop]Qwell[]: :D
20:04.15luke-jr_tainted-: "Please note: A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING TO WORK!"
20:04.17dlynesWhen I used to read Asterisk-Biz regularly, I swear probably 5-10% of the messages were about Nufone service :)
20:04.31dlynesluke-jr_: ztdummy will work, too
20:04.32Qwell[]dlynes: Now about 50% are about didx
20:04.38dlynesQwell: lol
20:04.40*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
20:04.46Qwell[]s/about/from/
20:04.49dlynesQwell: Yeah...what's his facer....gawd, that guy's annoying :)
20:04.52tainted-luke-jr_ lol n/m i was looking at the wrong box this whole time
20:04.56Qwell[]wtf
20:04.57Qwell[]stupid bot
20:04.58file[laptop]Rehan Allawallawhoha
20:05.04tainted-zaptel is required
20:05.10dlyness/about/from
20:05.25file[laptop]I liked his last post, about the hungry DIDs
20:05.31dlynesProbably the trailing forward slash screwed it up
20:05.39boddyhi I have one problem soft phones are in the same local network with asterisk asterisk service internet client behint the internet client can call softtphones but softphones doesnt call
20:05.40dlynesYeah...Rehan
20:05.45dlynesman that guy's annoying
20:05.49Qwell[]Who's that other idiot?
20:05.56Qwell[]I guess he only posts on -users...Douglas
20:06.03file[laptop]Douglas Garstang
20:06.13Qwell[]he's didx, right?
20:06.19tainted-u know who was a true idiot? ptg123 / preston garrison
20:06.22file[laptop]nah
20:06.24Qwell[]oh
20:06.30dlynesPreston Garrison?
20:06.36dlynesIsn't he the voipsupply guy?
20:06.37Delmardlynes, to hell with it. i need to upgrade the kernel and packages anyway.. ill update the box a whole heap and then see if its still broken and go from there.
20:06.40Qwell[]dlynes: no
20:06.48boddysoftphones register to  asterisk's local ip adres
20:07.00tainted-dlynes not sure.. he was just a con-artist
20:07.06dlynesah
20:07.11Qwell[]tainted-: Doesn't make him an idiot :p
20:07.15dlynesI recall seeing his name
20:07.21luke-jr_dlynes: ztdummy is still a driver
20:07.25boddy?
20:07.30tainted-luke-jr_ just install it
20:07.34dlynesluke-jr_: install it
20:07.36luke-jr_tainted-: no
20:07.39Ariel_luke-jr_, you need a timing source use it
20:07.51dlynesluke-jr_: It uses the rtc instead of a digium card
20:07.51luke-jr_Ariel_: /dev/rtc?
20:07.52tainted-luke-jr_ DO IT NOW
20:08.07luke-jr_dlynes: that should be user-mode
20:08.09afrosheenboddy: is * behind a firewall
20:08.15dlynesluke-jr_: it's not
20:08.19Ariel_luke-jr_, if you don't want to install the drive then fix the meetme not to use it.
20:08.19luke-jr_dlynes: should be
20:08.24dlynesluke-jr_: doesn't matter...it's not :)
20:08.28*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
20:08.34boddyafrosheen ?
20:08.41boddyafrosheen *
20:08.48dlynesluke-jr_: Not everything for asterisk is the way it's supposed to be...just get used to it
20:09.06dlynesluke-jr_: asterisk doesn't live in reality sometimes
20:09.13afrosheenboddy: asterisk = *, are you not a native english speaker?
20:09.16luke-jr_Ariel_: I'd be more likely to consider that if Digium would take GPL patches
20:09.28boddyyes
20:09.29*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
20:09.52dlynesDelmar: Well, if it's any help, Slackware 10.2 works out of the box with it :)
20:09.59Qwell[]luke-jr_: never gonna happen
20:10.07boddy* is behind the nat
20:10.17afrosheenboddy: so are the incoming calls sip or zaptel or what
20:10.47luke-jr_Ariel_: Qwell[]: or if someone does a proper fork of Asterisk
20:10.49boddysip
20:11.31afrosheenboddy: and are the incoming calls behind NAT also?
20:11.37Qwell[]luke-jr_: feel free..
20:11.42Ariel_wow one item at a time will take for ever to find out what is happening
20:11.55boddysip clients that on internet can register and call softphones
20:11.56luke-jr_Qwell[]: I don't have the time to maintain a fork on my own
20:12.05Qwell[]luke-jr_: Then deal with it. :)
20:12.07boddyyes soft phones behind the nat
20:12.29boddysharing same local ip address range with *
20:12.32afrosheenboddy: so you've got double nat, i.e.  outside sip phone > nat > internet > nat > asterisk?
20:12.53boddyyes
20:13.09luke-jr_Qwell[]: dealing with it involves not writing code for *
20:13.23boddybut sip phones can call softphones
20:13.29boddyno problem
20:13.30Qwell[]luke-jr_: Well, you aren't allowed to complain if something doesn't work then
20:13.41luke-jr_Qwell[]: hence my bug reports consisting of explanations on how to fix the bug without actual patches
20:13.43Ariel_externip = 204.123.456.100 localnet = 192.168.100.0/255.255.255.0
20:13.44boddysoftphones cant call
20:14.18boddyalso I have set extern ip and localnet
20:15.04timscottport forwarding
20:15.16timscottare you forwarding your RTP ports to your asterisk server?
20:15.46timscottboddy: forward your RTP ports to your asterisk server, and set your phones to always keep asterisk in the call path
20:15.51tasatare there any asterisk applications that will allow an agi to interrupt a wait?
20:15.59afrosheenboddy: I sugest you put * outside the firewall, i.e. dual home it with 2 network cards
20:16.10afrosheenboddy: that fixed ours perfectly
20:16.18timscottuh
20:16.39timscottyou don't need to move it outside the firewall, just forward your rtp ports.
20:16.40boddyafrosheen I will think this
20:17.03terrapenwoohoo the Redfone fonebridge works!
20:17.06terrapenyaaaay
20:17.06dlynesInfoworld was pretty cute today :)
20:17.15dlynesTaking revenge on end users:  http://newsletter.infoworld.com/t?ctl=11E4A32:41704E9
20:17.20terrapenthose guys rule.  awesome tech support
20:17.28blitzragecrack!
20:17.31*** join/#asterisk lzhang (n=rjrae@adsl-69-153-6-179.dsl.snantx.swbell.net)
20:18.05*** join/#asterisk redondos (n=redondos@190.48.54.2)
20:18.46lzhangI'm getting one way audio (polycom can't hear callee but callee can here polycom) on calls made to a certain number over zap
20:20.09*** join/#asterisk Maxxed (n=whyman@65.59.245.122)
20:20.15lzhangI'm doing a sip debug but I can't tell if there is anything wrong
20:20.26lzhanganybody want to take a look for me? I can pastebin
20:20.33Maxxedok! so, i got the chanspy working
20:20.43Maxxedbut i want to be quiet, and stick to a group
20:20.56Maxxedlike exten => 1337,1,ChanSpy(SIP|q,g(3333))
20:21.01dlyneslzhang: is there a firewall between the polycom and the asterisk box?
20:21.16lzhangno firewall, no NAT
20:21.28Maxxedhow do i use the q, and g at the same time ?
20:21.35Maxxedi know i know..
20:21.38Maxxedsilly..
20:21.38Maxxedheh
20:21.42lzhangasterisk box is the gateway and dhcp server as well, all phones are on the same subnet
20:21.52dlynesMaxxed: ChanSpy(SIP|qg(3333))
20:22.02Maxxedok
20:22.06Maxxedjust wanted to make sure
20:22.09Maxxedthx dlynes
20:22.10dlyneslzhang: asterisk box is the firewall?
20:22.14Maxxedim having one of thoooooose days..
20:22.15Maxxedheh
20:22.27lzhangdlynes: yes
20:22.48dlyneslzhang: Make sure that you don't have anything impeding udp connections on the firewall from the lan
20:23.01dlyneslzhang: i.e. check your iptables rules
20:23.11dlyneslzhang: If you have full logging enabled
20:23.18*** join/#asterisk zotz (n=zotz@24.231.32.85)
20:23.26dlyneslzhang: Then check your firewall log to make sure it's not junking packets that you think it's not touching
20:23.52lzhangdlynes: how can I turn on logging so I know if iptables is blocking the sip ports on the call?
20:23.55dlyneslzhang: I suspect you might have firewall issues stopping your rtp traffic from working correctly
20:23.58tasatDoes anyone here have any experience with using Playtones in an AGI?
20:24.22dlyneslzhang: Are you using raw iptables, or are you using some kind of external iptables configuration tool?
20:25.15lzhangdlynes: it's just an iptables script
20:25.28lzhangdlynes: will an iptables -L be helpful?
20:25.29*** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
20:25.36dlyneslzhang: Ok, so you're not using Redhat/CentOS/Fedora, then?
20:25.44dlyneslzhang: Yeah, that's helpful
20:26.06dlyneslzhang: logging rules will show .... -l if I remember correctly
20:26.06lzhangdlynes: I'm on debian sarge 2.6 kernel
20:26.21luke-jr_Qwell[]: would Digium take a BSD patch?
20:26.22dlyneslzhang: ok
20:26.27dlynesluke-jr_: FreeBSD?
20:26.32luke-jr_dlynes: BSD license
20:27.03dlynesluke-jr_: no idea...I know they do take copyrighted code....anthm's donated a lot of code to asterisk
20:27.07lzhangdlynes: http://pastebin.com/674048
20:27.47lzhangdlynes: thanks for your help so far, I haven't considered the possibility of the firewall blocking... this is on a certain phone number repeatedly though, and other numbers work
20:28.15dlyneslzhang: With firewall rules like that, you may as well not even be running a firewall
20:28.43Qwell[]luke-jr_: That's a question you should probably ask on licensing.digium.com
20:29.15ManxPowerYes, BSD licensed code can be put in Asterisk
20:29.35ManxPowersince BSD license does not prohibit usage in closed source or binary only products
20:30.03*** part/#asterisk miller7 (n=none@adsl19-22dynamic.athens.acn.gr)
20:30.04terrapenwhat code do you have, luke
20:30.10afrosheendlynes: lol
20:30.28ManxPowerin fact BSD licensed code is ALREADY in Asterisk
20:30.30dlynesafrosheen: well, did you look at the pastebin?
20:30.55dlynesafrosheen: sheesh...everything is set to default policy of accept, and every vulnerable port has been explicity set to accept, too :)
20:31.16afrosheenouch
20:31.22afrosheenorder accept, deny :)
20:31.24lzhangdlynes: haha I didn't write it, I'm just the one who ends up having to fix it
20:31.27dlyneslzhang: anyways...nothing i see on there that would stop it from working
20:31.45dlyneslzhang: it might be a problem specific to polycoms
20:31.58dlyneslzhang: what's the debug spew look like for it?
20:32.15dlyneslzhang: Are you serious a network/sysadmin wrote that script?
20:32.16afrosheenyeah we have a polycom specific problem..
20:32.52dlyneslzhang: That script pretty much leaves that machine wide open to the internet
20:33.17lzhangdlynes: please don't compromise my box! :)
20:33.58*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
20:34.06dlyneslzhang: anyways...needless to say whoever wrote that firewall script should be fired :)
20:34.30lzhangdlynes: I'm pretty sure it's polycom, I'm not knowledgable enough about sip to tell if the debug messages are good or bad
20:34.32tasatIs there a recommended way to play a 'ringing' tone to a user while waiting for a call origination (through manager) to complete?  I've been trying playtones before the originate action, and stopplaytones after -- but that doesn't do anything.  Seems like playtones needs a wait, and I don't know the wait period ahead of time.  I'd really appreciate any advice....
20:34.38lzhangdlynes: let me make a pastebin
20:35.06ManxPowertasat, Asterisk will provide ringing sound (called RINGBACK) to the caller by default.
20:35.20ManxPoweryou MAY need to have /etc/asterisk/indications.conf
20:35.33dlynestasat: .r option in the dial command, but asterisk will normally provide ringing...the ',r' will generate ringing where it may not be appropriate to do so
20:36.01tasatManxPower: it does when I do a Dial, but I'm using an orignate because the called user is placed into a conference
20:36.21ManxPowertasat, do you have /etc/asterisk/indications.conf ?
20:36.33*** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com)
20:37.00tasatManxPower: it's there...
20:38.03lzhangdlynes: http://pastebin.com/674070 so there's a sip debug
20:38.21*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
20:39.25*** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net)
20:39.33tasatManxPower: I don't suppose there's a wait to execute an indefinite wait, and then interrupt it?
20:39.42xmarkdoes anyone have an update on the nufone DID problem?
20:39.46lzhangdlynes: I think I'm about to try the line with just a regular phone
20:39.48tasatManxPower: from an agi
20:40.43dlyneslzhang: It looks like it might be a codec issue
20:40.46lzhangtasat: in agi, you can just do a while loop to wait
20:40.53*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
20:40.57dlyneslzhang: What's your sip.conf for the outgoing call look like?
20:41.11dlyneslzhang: Just the disallow and allow lines
20:41.51lzhangdlynes: wtf why are the allow disallow lines all commented out
20:42.00dlyneslzhang: you're asking me?
20:42.43Cybertoyanyone seen this ipcall2phone.com ?? the sip credentials they mailed me are not working ..
20:42.53Cybertoythey advertise free calls to brazil ...
20:42.56lzhangdlynes: haha no how can you tell from the sip debug about the codec issue? and many many thanks btw for your help so far
20:43.12dlyneslzhang: if you look at the end of your log
20:43.22dlyneslzhang: You'll see that the remote end was only supporting ulaw/alaw
20:43.34dlyneslzhang: Which is the default codecs supported under asterisk
20:44.16LostFrogIs there a decent WiFi VoIP phone?
20:45.27Qwell[]That pastebin isn't a codec problem.
20:45.27*** join/#asterisk SoMeOnEnUlL (n=morris@p726-adslbkkct1.C.csloxinfo.net)
20:46.00xmarkanyone else affected by the nufone DID issue?
20:46.07Qwell[]xmark: everybody
20:46.13*** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net)
20:46.14bigblock02yeah, no help will come from this channel though
20:46.27bigblock02im porting my number away
20:46.44Qwell[]bigblock02: k, shh now
20:46.48dlynesQwell: One side's offering up ulaw/alaw/g729; the other side is offering up ulaw/alaw
20:47.01Qwell[]dlynes: yes, and it'll allow ulaw or alaw
20:47.02Zodiacalqwell still want me to test your skinny?
20:47.03dlynesQwell: I suspect his upstream doesn't use ulaw or alaw
20:47.03bigblock02why?  cause you are not?
20:47.11bigblock02im stating the fact thank you
20:47.13Qwell[]dlynes: His upstream is zap
20:47.24dlynesQwell: Then why is he going out on sip?
20:47.31*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
20:47.53Qwell[]-- Executing Dial("SIP/8002-d5ef", "Zap/r1/6491306||wW") in new stack
20:47.57Qwell[]looks like zap to me...
20:48.04dlynesYeah, but he's receiving another call
20:48.09dlynesAnd it's coming in on sip
20:49.06Qwell[]Which call is failing?  I see a few here
20:49.06bigblock02nufone website has been updated...
20:49.06b00mer_When I do a call forward on my phone the caller ID isn't passed along, is there a way to fix that other than making asterisk do the forward?
20:49.07Qwell[]file[laptop]: ^^
20:49.07ManxPowerb00mer_, it does that by default
20:49.07ManxPowerat least in 1.2
20:49.15tainted-bigblock02 updated to what?
20:49.17bigblock02under the press tab
20:49.30bigblock02heads up on the DID issue
20:49.31b00mer_ManxPower: it doesn't pass it by default?  How do I enable it?
20:49.36Qwell[]SIP/2.0 407 Proxy Authentication Required
20:49.43ManxPowerb00mer_, it passes it by default
20:49.47Qwell[]dlynes: That isn't a codec thing at all
20:50.05dlynesQwell: ah...didn't notice there was three calls going on there
20:50.05b00mer_ManxPower: yea when I do the forward on the phone its not :(
20:50.07bigblock02looks like if you have a MI number you are out of luck, TF will be ported but i doubt their eta is accurate
20:50.17dlynesQwell: I missed the other dial
20:50.26Qwell[]bigblock02: #nufone
20:50.52bigblock02right..
20:50.57lzhanguplines is definately zap
20:51.11bigblock02no one is talking in nufone about this
20:51.40Qwell[]bigblock02: Nobody is talking in here about it either.
20:51.43file[laptop]that doesn't mean you can talk about it here
20:52.05lzhangQwell[]: is it a proxy problem?
20:52.11Zodiacalanyone setup the AstTapi outlook dialer with sccp? it seems to lockup outlook for me... i got it to work once, but stupid me didn't write down the config settings...
20:52.13Qwell[]lzhang: authentication problem, from what I can tell
20:52.15bigblock02do you guys work for them or something?  i dont get it... its not like im out of control or anything
20:52.30file[laptop]no, I don't
20:52.30bigblock02there was all this talk about the weather a while back and that was ok?
20:52.40Qwell[]bigblock02: yes
20:52.44dlynesbigblock02: No, but how are we going to help you?  We're not
20:52.49bigblock02how is the weather more on topic than talk about AN ASTERISK COMPANY?
20:52.49tainted-bigblock02 lol, let me know if u get anywhere with that argument
20:53.01Qwell[]bigblock02: nufone IS NOT an asterisk company
20:53.07bigblock02i beg to differ
20:53.12dlynesbigblock02: People aren't expecting us to change the weather...you're expecting people to help you fix your nufone problem
20:53.17file[laptop]they use Asterisk, doesn't mean their stuff should be discussed in #asterisk
20:53.19bigblock02their switches run asterisk, correct?
20:53.25bigblock02the CEO is IN THIS CHANNEL, correct?
20:53.37dlynesbigblock02: so /msg him
20:53.46dlynesbigblock02: email him
20:53.49luke-jr_bigblock02: I doubt it, really
20:53.49Qwell[]bigblock02: Go bitch in #ubuntu to the CEO of redhat
20:53.59bigblock02weather is good over here... hows it out there?
20:54.04Ariel_folks there is ignore on the channel
20:54.10Qwell[]Ariel_: There is also +q
20:54.10dlynesGreat...nice and sunshiny
20:54.11bigblock02partly coudy... looks like it may rain
20:54.16blitzrage#asterisk is not a VSP support channel
20:54.22Qwell[]blitzrage: VSP?
20:54.23file[laptop]sucky here, looks good out but it's cold :\
20:54.24Qwell[]nm
20:54.29blitzrageQwell[]: heh :)
20:54.36Qwell[]silly canadians and their ackward terms :p
20:54.40Qwell[]It's ITSP! :p
20:54.41Ariel_hot and dry outside
20:54.47LostFrogNo recommendation on a WiFi hard phone?
20:54.50tainted-hey if nufone is looking for termination, i can help him out
20:54.59ManxPowerI give up
20:55.00blitzrageLostFrog: hitachi
20:55.01*** part/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
20:55.01b00mer_how about the call forwarding issue not passing caller id when forwarded from the phone... I assume its different than when I would do it from a dialplan? right?
20:55.10Ariel_LostFrog, I have a utstarcom F1000 which is small but works fine
20:55.14Nugget<Tech Support> To abort you need to hit ^A     <Baffled Canadian> I'm hitting control, eh?  Nothing's happening.
20:55.18Qwell[]LostFrog: The new cisco one looks nice
20:55.25blitzrageb00mer_: look at the 'o' flag for Dial()
20:55.25Qwell[]erm, linksys, heh
20:55.38macTijnNugget: lol
20:55.40*** join/#asterisk jlewis (i=jlewis@solo.atlantic.net)
20:55.44bigblock02i have a problem... whenever someone dials my TF number they get an busy signal... any ideas?
20:55.47macTijn<- ZzZz
20:55.50bigblock02lol
20:55.54Qwell[]file[laptop]: ?
20:55.55LostFrogHas anyone used the linksys phone with *?
20:56.03file[laptop]OKAY, I need to make a phone call
20:56.08bigblock02my asterisk configs are the same as they were yesterday
20:56.11Qwell[]file[laptop]: Can we get a little +q please?
20:56.14tainted-bigblock02 lol!
20:56.21luke-jr_LostFrog: linksys has a phone?
20:56.22dlynesbigblock02: Could be your end, could be nufone's end, could be nufone's pstn provider's end
20:56.34luke-jr_maybe the PSTN is down!
20:56.37bigblock02hrmm... maybe i'll call nufone
20:56.40Ariel_funny my tf number just rings
20:56.41blitzragethe internet is down!
20:56.42tainted-omg this is the best comedy i've seen in a long time
20:56.56jlewisanyone know anything about a "new" g729 key resetting policy at digium?
20:56.57b00mer_blitzrage: saw the 'o' flag, but is that how it works on the phone too?
20:56.59*** join/#asterisk MikeJ__ (n=vircuser@64.241.37.140)
20:57.05b00mer_the internet is down??? call Al Gore!
20:57.07*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
20:57.09luke-jr_when my TF number doesn't work, it won't even ring... just silence on the callee's end
20:57.09dlynesjlewis: what new policy?
20:57.10*** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za)
20:57.10Qwell[]jlewis: never heard of it...you'll have to ask them
20:57.13jlewis"3 registrations and your're done"
20:57.17Qwell[]heh, bad timing
20:57.22jlewisthey just told me
20:57.28jlewisI'm kind of pissed
20:57.29*** mode/#asterisk [+o file] by twisted[asteria]
20:57.29Qwell[]once bookshire goes down...they all do
20:57.30dlynesjlewis: When I talked to them about two weeks ago, they never mentioned anything about it
20:57.35Qwell[]jlewis: well, explain it to us?
20:57.35*** mode/#asterisk [+o file[laptop]] by twisted[asteria]
20:57.38*** join/#asterisk X-Gen (n=x-gen@dsl-145-233-158.telkomadsl.co.za)
20:57.42jlewismaybe the support just was FoS
20:57.44lzhangI'm not sure if it's a Polycom issue anymore, I just tried dialing the number through Xlite and I'm getting the same result
20:57.44Qwell[]twisted[asteria]: nub
20:57.49LostFrogluke-jr_: yes.. WIP330
20:57.50file[laptop]now, as we were saying...
20:57.58luke-jr_LostFrog: Work In Progress?
20:58.17Ariel_lzhang, xlite does not have g729 pro does
20:58.44lzhangAriel_: trying to dial out over Zap using ulaw
20:58.59Qwell[]dlynes: See, I told you the upstream was zap :p
20:59.06luke-jr_um... isn't Zap analog?
20:59.14file[laptop]Zap can be many things
20:59.18jlewisQwell[]: what's to explain?  I frequently have to move g729 keys when customers upgrade servers or we have to move customers to temp servers due to hardware failures
20:59.20dlyneslzhang: Qwell suggested the problem might be your proxy settings...did you see that?
20:59.38Qwell[]lzhang: It would also be incredibly helpful if you only had it show one call, and not 3-4
20:59.38jlewisnow, digium is telling me I can't do this anymore without re-buying the g729 keys
20:59.46Qwell[]jlewis: huh
20:59.47lzhangdlynes: Yes, and I'm trying to find out what that means right now. SIP proxy settings? where do I see that
21:00.05dlyneslzhang: Did you configure an outbound proxy for the polycom?
21:00.07Qwell[]lzhang: It looked like an auth thing
21:00.13jlewishe claimed it's been policy for several months...but it's not in their current policy on the web site
21:00.18file[laptop]jlewis: have you done it many times with the same keys?
21:00.36dlyneslzhang: also make sure you've got the correct password and username set
21:00.36jlewisI wouldn't say many times with same key...but many times with different keys
21:00.50jlewiswe've got probably a few $k worth of keys
21:01.33LostFrogThe WIP330 only supports Ulaw and G729a.
21:01.43dlynesjlewis: I just know that after the second time, you have to explicitly get permission from the accounting department or something before you can get it re-registered on another device
21:01.58jlewisdlynes: yeah...I've done that many times
21:02.15b00mer_blitzrage: saw the 'o' flag, but is that how it works on the phone too?  When I forward the call ... I get unknown number on the forwarded to phone.
21:02.17jlewisany time I move a key, I email support@digium explaining why it was moved and asking for a reset
21:02.26lzhangdlynes, Qwell[]: http://pastebin.com/674105 That should be just one call
21:02.47jlewisthey ignored my last request, and now say I can't do this anymore
21:02.47b00mer_blitzrage: I added the 'o' option to the initial extension receiving the call...
21:02.58b00mer_blitzrage, but I can't see how that helps
21:03.05file[laptop]I'm not in that department sadly enough so I don't know what's up... lemme see here
21:03.23lzhangjust to be clear guys, I'm having this problem intermittently, the phones can make calls to other numbers over zap fine, I'm trying to debug this number because I can get it to reliably produce the problem
21:03.35jlewisbased on hold times, I'm suspecting digium's closed...but the phone queues don't know it
21:03.45brodiemanyone recommend a book other than TFOT? maybe something that includes more about protocols/telephony in general? long flight is coming near..:)
21:03.46jlewisI think I'll leave it on hold over the weekend on their dime
21:03.47fileclosed? no
21:04.02tainted-any ideas on preventing a sip peer from timing out?
21:04.14Qwell[]lzhang: And what actually happens?
21:04.29tainted-i've got stun up, qualify = 5000 but shit still times out
21:05.41b00mer_anybody else know how / why my phone would not pass along the callerid when forwarding.  Its a function of the phone right?  not asterisk?
21:06.51jsharptainted-:  If you find out, let me know.  I have the same problem on phones that are on the same FastE hub as my * box.
21:06.54*** join/#asterisk denon (i=denon@synapse.subneural.net)
21:06.54*** mode/#asterisk [+o denon] by ChanServ
21:07.14tainted-LOL
21:07.28jsharpFastE switch, rather.
21:07.34tainted-800-807-4515
21:07.45dlyneslzhang: I don't see any problems with it, from the debug log this time, myself
21:07.48tainted-'if u are a nufone customer yada yada yada'
21:08.08tainted-jsharp yea it's been isolated to one specific person
21:11.06brad_msswterrapen: around ??
21:12.33*** join/#asterisk Craziman2 (n=Craziman@110-host76.planetc.com)
21:13.32*** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com)
21:13.39jlewisback on phone with digium...he admits he sees nothing about this policy on the web site...I'm on hold and he's "checking with legal"
21:14.06filetold you we weren't closed ^_^
21:14.23mutilatorO_o
21:14.24mutilatorwhat'de i miss?
21:14.27jlewistook much longer to get through this time
21:14.29mutilator"checking with legal"
21:14.55*** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com)
21:14.58jlewisI think he thinks I'm "pirating" g729 or something
21:15.54*** join/#asterisk copantl (n=galellop@63.245.93.138)
21:16.31copantlhow can i configure peers authenticating by prefix?
21:16.40copantlwith sip
21:17.09JerJer(17:16:35) tainted-: dude i want to beat u down at the next convention
21:17.09JerJer(17:16:44) tainted-: u won't know who it is and where it's coming from
21:17.09JerJer(17:16:48) JerJer: because you do not know the situation
21:17.09JerJer(17:16:53) tainted-: i'm going to smack u shitless
21:17.09JerJer(17:16:55) JerJer: go ahead and try it
21:17.09JerJer(17:16:55) tainted-: u fucking wait
21:17.32[hC]Hah
21:17.32terrapenerr on
21:17.36[hC]Nice jer.
21:17.38[hC]What the hell.
21:18.49jlewistesting my patience is more like it
21:19.09*** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com)
21:19.41distortionI <3 telecom
21:19.54Qwell[]telecom </3
21:20.17*** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com)
21:20.29tainted-JerJer stop trolling haha
21:22.07[hC]tainted-: im not sure if youve noticed, but jer is a fairly big guy.. you may wanna be careful with what you threaten.. :/
21:22.25Qwell[][hC]: Was waiting for somebody to mention that :p
21:22.52terrapensweeeeeeeet I got my Utah personalized license plate
21:22.55[hC]I could see it now.. tainted running around swinging and cursing while jer simply bonks him over the head with a fist.
21:23.06tainted-hahaha
21:23.16terrapenTELEMRK   (skiing, not telemarketing!)
21:23.21jlewisanyone ever get that g729 code based on intel's libs working?
21:23.28tainted-i'm a pacifist
21:23.35tainted-he's just pasting random stuff
21:23.36Qwell[]jlewis: yes, but it isn't legal to use
21:23.49JerJertainted-:  everybody in #Nufone saw what you said
21:23.53*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
21:24.19tainted-JerJer there's like 5 of us laughing at your reaction dude
21:24.41tainted-i don't do that stuff
21:25.00tainted-anyways, keep it in #nufone
21:25.26*** join/#asterisk themikester60 (n=mikey@66-100-35-23-static.dsl.oplink.net)
21:25.36JerJerthen don't threaten to attack me physically for a situation that is totally out of my hands at this moment in time
21:26.08tainted-i'm like 12lbs
21:26.11*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
21:26.21tainted-and that's while carrying a 1U box
21:26.22themikester60alright everyone, I've got a very interesting scenario for you, suppose you want to use the g726 codec with asterisk and you've set all your phones to accept only that codec. Now assume you have remote phones that are behind NAT and they can place calls out but if you try to call their extension it doesn't work at all, any ideas on how to go about fixing that?
21:26.31b00mer_lets all relax and ponder my issue... how do I get call forwarding on my 7960 to pass the callerid
21:26.54JerJerb00mer_:  i have not seen anything that lets you do that
21:27.12b00mer_JerJer: thanks!  ok... back to the drawing board
21:29.41brodiemthemikester60, did you set externip in sip.conf?
21:29.48tainted-JerJer: i'm going to puncture your lung with my Bic
21:29.48tainted-tainted-: dude let it go - just a joke
21:29.48tainted-JerJer: you think it's funny? things are going up in flames and no one wants to buy me out!
21:29.48tainted-tainted-: man that really sucks.. why'd telestetic dump u on your ass?
21:29.48tainted-JerJer: some debt disputes - now fuck off!
21:30.27JerJeras you well know those are not my words
21:30.28brodiemthemikester60, my bad n/m that
21:30.32themikester60brodiem, no I havn't, and in fact I'm not entirely sure I've ever heard of that
21:30.39themikester60thanks anyway brodiem
21:30.48brodiemthemikester60, yeah that was for ast running behind nat not the other way around
21:31.07brodiemthemikester60, your sip extensions type=friend?
21:31.24themikester60yup
21:31.51tainted-JerJer everyone in #fuckedcompany saw what u typed dude
21:31.57b00mer_anybody notice is asterisk sounds there is an on.gsm but no off.gsm ?
21:32.01JerJertainted-: prove it
21:32.14*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com)
21:32.18Math`anyone familiar with Nortel Meridians? Im trying to connect a PRI from an M1 to a TE110P, signalling goes through fine, but the call immediatly hangs up
21:32.19file[laptop]don't make me kick you two
21:32.24Math`here is a pri debug span 1: http://pastebin.ca/50339
21:32.27[hC]of course they did
21:32.33terrapenwho the fuck cares, guys?
21:32.34[hC]theres only one person in #fuckedcompany
21:32.34Maxxedim trying to get a varaible set if a caller id value starts with 21, cant this be done with an if statement?
21:32.36Maxxedexten => s,1,SetVar(SPYGROUP=SalesSpyGroup1{IF(${CALLERIDNUM} = 21xx))
21:32.37[hC]TAINTED.
21:32.44tainted-i klined them all
21:32.51tainted-nufone naysayers
21:32.53Maxxedcaller id is the source ext
21:32.57Maxxed4 digits
21:32.58terrapenshut up about nufone
21:33.11Maxxednot sure how to use it in the extentions.conf
21:33.25copantlhow can i authenticated with a prefix in SIP
21:33.28copantlany idea?
21:33.38bkw__copantl, that usually means adding a prefix to the number
21:33.41bkw__ie 1234
21:33.49Math`copantl: put the user in a context which has nothing else but that prefix
21:33.51bkw__then you dial SIP/1234${EXTEN}
21:34.05brodiemthemikester60, are your sip phones registering? sip show peers
21:34.06tainted-http://www.sineapps.com/images/jerjer.jpg <-- "where's all my customers going?"
21:34.09copantlis for terminaion
21:34.29copantlis not only a call
21:34.31Maxxedanyone know how to use IF in the extentions.conf ?
21:34.39*** kick/#asterisk [tainted-!n=jcolp@mctnnbsa24w-142167058071.pppoe-dynamic.nb.aliant.net] by file[laptop] (file[laptop])
21:34.39*** join/#asterisk tainted- (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net)
21:34.39Math`Maxxed: check application  GotoIf
21:34.44terrapenare there any voip-info.org admins around here?
21:34.45Maxxedwell..
21:34.48tainted-file[laptop] sorry i'll stop
21:34.52brodiemthemikester60, also enable sip debugging
21:34.55Maxxedgoto if will work, but i rather one line it
21:34.57copantlMagh`
21:35.05themikester60brodiem, yea the phones are registered and work fine, and I can place calls outward, but you can't call in and reach the phones, however when g726 is turned off as the forced codec then the calls work both incoming and outgoing
21:35.17terrapeni have a few gripes about voip-info.org, namely the automatic editing when you click somewhere on a wiki page...and the fact that searches open a new window
21:35.21jlewisQwell[]: I suspect it is legal...if you've already paid digium the license/royalty fees
21:35.35jlewisI just got done talking to digium legal
21:35.35Qwell[]jlewis: The intel g729?  no
21:35.36brodiemthemikester60 oh didn't realize only 726 was failing
21:35.37copantli wan to create a peer  with authentication by a prefix
21:35.40copantlnot a user
21:35.52themikester60brodiem, yea thats what makes this quite the odd scenario
21:35.52Qwell[]jlewis: Nobody is allowed to use it...it's a teaching tool only
21:36.04jlewisapparently they're cracking down on g729 reregistrations because they feel some 'customers' have been abusing it and stealing licesnses
21:36.05JerJeruser/peer on sip is not as defined as it is on iax
21:36.11Math`Qwell[]: there's probably a way of buying a license to use it tho
21:36.19brodiemthemikester60, did you allow=g726 in your sip.conf?
21:36.23Qwell[]Math`: I don't think there is..
21:36.29jlewisI just need to ensure that we can use what we've paid for
21:36.32themikester60brodiem, yea and I did disallow=all right before it
21:36.41Qwell[]jlewis: Stop changing hardware :p
21:36.41Math`Qwell[]: then how did digium get the license... they're paying someone for it
21:36.50Qwell[]Math`: no, the intel g729 license.
21:36.50jlewisthats not an option
21:36.53Qwell[]erm, code
21:37.02jlewismotherboards die, customers want faster boxes, etc.
21:37.06Math`oh
21:37.17Maxxeddamn, how would i only get (${CALLERIDNUM} = 21xx))
21:37.25Maxxedlike anything that starts with 21
21:37.26brodiemthemikester60, same with the contexts specific for the actual extensions?
21:37.40Maxxed2112, 2100, 2188 blahbityblahblah
21:37.42jlewismaybe they should to back to a LPT port dongle or something
21:38.01jlewistying the key to a particular motherboard kind of sucks
21:38.20themikester60brodiem, yea I configured it there as well
21:38.24Zodiacalqwell know how i can change the sccp ringtone dymanicly in my dialplan? i used to be able to do this with SIP using _ALERT_INFO but it doesn't seem tow ork with sccp
21:38.27LostFrogUSB dongle.
21:38.39Zodiacaldymanicly = dynamicly :P
21:39.13brodiemthemikester60, i'd turn on sip debugging then
21:39.33themikester60brodiem, I've turned it on but so far have yet to find anything useful in the logs, I keep getting a sip error 488
21:40.14*** join/#asterisk pingywon (n=mike@c-69-242-23-135.hsd1.pa.comcast.net)
21:40.25redondosI would like to perform a Goto() when the user hits the * in his phone. How would the extension look like? Is it just: exten => *,1,Goto(whatever) ?
21:41.46brodiemthemikester60, canreinvite=no in sip.conf?
21:42.01themikester60brodiem, yea its set that way
21:42.41brodiemthemikester60 what kind of phones are you using
21:43.03themikester60I'm using a snom 320 and a spa941
21:43.27*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
21:45.17brodiemthemikester60, hmm seems odd to me.. i have a bunch of 360's and never ran into any codec negotiation issues like that
21:45.33Maxxedman.. wtf this is lame
21:45.34brodiemthemikester60, try allow=all and comment out the disallow=all just to see if it'll neg. with 726 then
21:45.34Maxxedheh
21:45.45themikester60brodiem, alright let me give that a shot
21:45.45redondosAnyone?
21:45.48b00mer_i understand dbget is deprecated... will a set do a +101 if db value not found?
21:46.10Maxxedexten => s,1,SetVar(SPYGROUP=SalesSpyGroup1{IF(${CALLERIDNUM} = 21xx)) .. anybody know what im goin for here?
21:47.05Nuggetan aneurysm.
21:47.17wunderkinMaxxed, /usr/src/asterisk/doc/README.variables
21:47.56*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
21:48.32Maxxedwunderkin: thanks again :)
21:48.34themikester60brodiem, I tried what you said and I can place a call from the snom to the sipura and the snom is in g726 format whereas the sipura is in ulaw, but if I try to call from the sipura to the snom it will ring but then just hang up when I answer the snom
21:48.48*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
21:48.52Maxxedwunderkin: i normaly just need a point in the right dirction
21:49.01Hmmhesayswow what a trainwreck this script is
21:49.04Hmmhesaysbut it works
21:49.07Hmmhesaysso i'm happy
21:49.22themikester60brodiem, nix that, now both of them hang up when you answer one
21:50.49nextimehi, what can produce an HDLC bad FCS (8) on a pri (t410p) other than interrups problems?
21:51.28*** join/#asterisk kEpEx5 (n=kepex@dsl-200-67-17-114.prod-empresarial.com.mx)
21:51.35nextimeit's specific to only one of the 4 spans on the card, others work without problems
21:51.42nextimemaybe a bad cabling?
21:51.55*** join/#asterisk Olobola (n=casper_s@adsl-75-6-237-183.dsl.pltn13.sbcglobal.net)
21:52.32tdonahuehi all
21:52.33brodiemthemikester60, can you dial a local asterisk extension from both phones and neg. 726?
21:53.03tdonahuecan the configuration for snom phones be done from a ftp/tftp server or do I have to use the web interface?
21:53.41swytchtdonahue: hi
21:53.58swytchtdonahue: im sorry not to know you tho..  8-)
21:54.40tdonahuei had that nightmare last night...
21:55.25tdonahueweird thing is it became a reality today...
21:55.28themikester60brodiem, yea they can call between each other, I'm starting to wonder if perhaps you can't have two concurrent uses of g726?
21:55.30kEpEx5hello i have a asterisk box in panama, with a E1 ISDN PRI euroisdn, my problem is when i want to play a sound before anwer the call... and when i put playback(soundfile,noanwer) the soundfile not hears..
21:56.32*** join/#asterisk |cleric| (n=dacleric@p54829541.dip0.t-ipconnect.de)
21:56.58nextimekEpEx5 : you can't play a file before to answer. You can do Anser before.
21:58.24tdonahuei found the option i was looking for on the 360... now time to find the documentation for it...
21:58.28kEpEx5this is because the signaling PRI right?
21:58.39LostFrogtdonahue: what option?
21:58.59LostFrogmass configuration is done via http.
21:59.07nextimekEpEx5 : yes, the noanswer option is only for some channels, for zap pri channels it don't work.
22:02.03kEpEx5nextime  : ok tanks very much
22:03.34tdonahueLostFrog: i found the "Setting URL"  option, i just need to sit down and read about how it works.
22:04.12*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net)
22:04.28LostFrogI find setting DHCP option tftp-server-name works baest.
22:04.35LostFrogbest, even.
22:04.59tdonahueon the snoms? i didn't think they could do TFTP...
22:05.03LostFrogI use: option tftp-server-name "http://192.168.22.1/cgi-bin/snomcfg.pl?MAC={mac}";
22:05.08tdonahueahhh
22:05.38LostFrogIf I could get sipsak working, I would use the settings URL.
22:06.24tdonahueunfortunately we have multiple types of phones on the network, snoms, polycoms, ciscos...
22:07.05tdonahueso i have no idea how we are going to make them all pull their configs yet if they all take the tftp-server-name option
22:07.11[hC]anyone had any experience playing with win32 based call managers? I tried ADM but it seems to be about 70% functional
22:08.21LostFrogtdonahue: use class matching in your DHCP server.
22:08.23wunderkintdonahue, me too!
22:09.26tdonahuei'll take a look at it
22:10.00LostFrogAll snom phone macs start with 00:04:13
22:10.05*** part/#asterisk kEpEx5 (n=kepex@dsl-200-67-17-114.prod-empresarial.com.mx)
22:10.24tdonahuecisco has 3 manufacturer ids that i know of at this point
22:10.29austinnichols101tdonahue better get used to a world where everything "Almost Works" (tm)
22:11.31tdonahueaustinnichols101: one can always dream you know....
22:11.52austinnichols101disney phone
22:12.21tdonahueaustinnichols101: with the mickey mouse ears on the handset?
22:12.35austinnichols101and perfect sound quality
22:15.35*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-131-176.gdrpmi.dsl-w.verizon.net)
22:17.40bkw__VAD IS NOT BAD
22:17.45bkw__say it with me fellas
22:17.46*** join/#asterisk hydride (n=nathan@HSE-Montreal-ppp140758.sympatico.ca)
22:17.54LostFrogIt is with *.
22:17.54hydrideis asterisk@home a whole linux distrobution?
22:18.05bkw__LostFrog, well not really
22:18.11LostFrog~freepbx
22:18.16X-Robhydride, it's CentOS 4.3 (which is, yes, linux), with * and freePBX
22:18.37hydrideis there anyway I can get the features of it without having to install centos?
22:18.37LostFrogIs freepbx any better than AMP was?
22:18.46X-RobLostFrog, shitloads.
22:18.54hydrideor is there a nice webinterface I can use with debian?
22:18.56bkw__asterisk can't support VAD in the terms of transmitting/non-transmittal of packets when someone isn't speaking
22:18.58X-Robhydride, yes. Install it yourself.
22:18.59bkw__which sucks
22:19.11X-Robbkw__, is that still true?
22:19.16X-Robcoz trunk has it's own sip timer now
22:19.31tdonahuewhat the hell does VAD stand for?
22:19.42bkw__X-Rob, last I checked yes
22:19.51LostFrogtdonahue: Silence suppression.
22:19.54LostFroglol
22:20.02X-Robbkw__, I thought the idea of having the timer in there was to resolve those stupidisms
22:20.15tdonahuehmm... VAD == Silence Supression... yes i can see that now....
22:20.22LostFrogVoice Activity Detection
22:20.26X-Robhydride, freePBX is a nice gui. I am biassed, I'm a developer of it.
22:20.36tdonahueLostFrog: ahh, that makes more sense
22:21.44austinnichols101hydride, freepbx is excellent
22:21.50LostFrogWish I could use it.. our lines spend a lot of time on hold and on idel.
22:22.01LostFrogidle, even.
22:22.53hydridethanks
22:23.12tdonahuewell, i've had it for the night time to go home
22:23.20tdonahuehave a good weekend all
22:23.37Drukenoh, freepbx is just amp...
22:27.57*** join/#asterisk |omni| (i=cathode@216.64.178.146)
22:31.54*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
22:32.49generalhanwhats up all
22:35.13Hmmhesaysthe sky
22:35.20generalhani see
22:35.21generalhanlol
22:35.53Hmmhesaysbank of america and their stupid fscking online banking rules
22:36.04X-RobDruken, yeah. Well. I think there's only about 10% code left from AMP 1.10 to freepbx 2.1
22:36.10generalhanhmmhesays: i have an off topic question for you. i need to put in a firewall/router box and im sure youve seen me in here enough to know my lack of Linux knowledge ... can you just make a suggestion to me ?
22:36.12X-Robit's all modular and pluggable and extendable.
22:36.13*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
22:36.31*** join/#asterisk bavibs (n=jj0@c-67-166-96-37.hsd1.ut.comcast.net)
22:36.41Hmmhesayssave yourself the trouble and go buy a linksys?
22:37.17X-RobHmmhesays, you shouldn't be recommending linksys stuff - the WRT54's are no longer linux based.
22:37.37timscottso? who cares, as long as they work.
22:37.38generalhanwell i have a Linksys router, and i can control some things that way ... but a lot of the employees here have been abusing their internet access and my boss wants me to shut them down ... block ALL website domains except the one that they need to do their jobs
22:37.41X-Rob(not invalidating the point, they are good, but you still shouldn't be recommending them)
22:37.53HmmhesaysX-Rob, i didn't say what linksys
22:38.00Hmmhesaysor make any mention of linux
22:38.03X-Robgeneralhan, that's easy - quit.
22:38.15generalhanhaha yea i wish it was that easy
22:38.16Hmmhesaysor any mention of using linux on a linksys
22:38.28Hmmhesaysif I had, i would have said get the wrt54gl
22:38.36X-RobHmmhesays, linksys == bad, in general, because the WRT54's are no longer linux based.
22:38.49HmmhesaysYou need to read more dude
22:38.56X-Rob???
22:39.01X-RobHmmhesays, maybe you're missing the point
22:39.02X-Robit's a protest
22:39.05X-Robdon't recommend linksys
22:39.15X-Robuntil they sell decent wrt54's again
22:39.16Hmmhesaysafter linksys turned the wrt54g into a vxworks box, they released the wrt54gl
22:39.22generalhanwell i was going to use an opensource project like Smoothwall cause of the nice web interface, but you HAVE to have IDE Hard Disks which i dont have in my server.
22:39.31Hmmhesayswhich DOES run linux
22:39.33X-Robyes, which has less RAM and flash than the old v4 wrt54's
22:39.38X-Robuh v3's
22:39.59X-Roband it's still more expensive
22:39.59Hmmhesaysit has the same amount of ram and flash as the original wrt54g's
22:40.18X-RobHmmhesays, 'original'?
22:40.22X-Robwhich 'original'?
22:40.26Hmmhesaysv1's v2's
22:40.30X-Robso 4/8
22:40.33X-Robwhich is crap
22:40.39X-Rob(as I said at the start)
22:40.42Hmmhesaysyou just want to whine about something
22:40.53X-RobNo, I'm saying don't support linksys
22:41.22X-Robdraytek are great, but I don't know if you can get 'em in the US
22:41.22Hmmhesaysand it's 4/16
22:41.49X-RobAsus WL-500g Deluxe are what I'm using these days
22:41.53X-Rob200mhz CPU, too.
22:41.58X-Robplus USB port
22:43.02X-Rob32m RAM.
22:44.21X-RobGSv3 was the best 8-\
22:44.23Shaun2222whats so special about the wrt54G version 1's? i have the v1.1 (upgraded to 4 now or somthing)?
22:44.54Hmmhesayshardware revision, not software
22:45.11Hmmhesaysanyway, i've had enough tech for the week
22:45.17Hmmhesaysi'm gunna go get drunk now
22:45.20timscottlater.
22:45.24generalhanlol
22:45.37*** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net)
22:45.52generalhani still need a good opensource firewall software ....
22:46.01Shaun2222Hmmhesays: the sticker on teh bottom is 1.1...
22:46.01tier_1pf
22:46.03timscottiptables! :D
22:46.05tier_1or ipf
22:46.14generalhani use iptables now
22:46.20timscottoh :(
22:46.35Hmmhesaysi wrote a nice cli proggy for loaded blocked ip's from a text file
22:46.42Hmmhesaysi lied, i'm waiting for my pizza
22:46.48timscott:p
22:46.53generalhanim looking for something that will allow me to lock down my users. my boss is pissed off at the ammount of playing on the internet by the employees so he wants me to lock them out of all domains but our one hosted software site
22:47.21Hmmhesaysso use iptables
22:47.38Netgeeksor use ACLs on your router
22:47.50generalhani dont have that kind of router ! lol
22:48.04generalhanif i had some Cisco router i wouldnt even be asking that i could deal with
22:48.13*** join/#asterisk MikeJ[Laptop] (n=vircuser@m810f36d0.tmodns.net)
22:48.20generalhanmy boss is cheap and wants FREE. so thats what i have to research
22:48.46Hmmhesaysjust drop everything thats not the ip of your domain
22:48.47Netgeekswell, you could do what Hmm said... take a low end pc, put linux on it, two interfaces and use iptables and ip forwarding
22:49.17generalhani guess i could do that ... and just move the server infront of all the net switches
22:49.27b00mer_any recommendations for a professional speaking auto attendent?
22:49.33b00mer_not computer
22:49.34Netgeeksyes, between the router and everyone
22:49.41timscott/usr/sbin/iptables -A INPUT -i eth0 -p all -s ! 192.168.0./4 -j DROP
22:49.54b00mer_are there websites that offer copy reads?
22:50.12generalhannow i would have to put the allow statement in for our off site hosting company first right ? then the drop all ?
22:50.20b00mer_like someone to read by message?
22:50.53Netgeeksgeneral good iptables and config there are some great docs out there on exactly what you need to do
22:51.19Netgeeksgoogle itables and config I meant
22:51.32generalhani figured thats what you meant lol
22:52.32*** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com)
23:01.00*** join/#asterisk Maxxed (n=whyman@65.59.245.122)
23:01.04Maxxedok.. back again
23:01.05Maxxedheh
23:01.07Maxxedso uh..
23:01.09MaxxedGotoIf($[${CALLERIDNUM} = 26${EXTEN:2}]?3:5)
23:01.11Maxxedthats wrong
23:01.35Maxxedany extention from 2600-2699 will go to 3
23:01.47Maxxedwell.. thats the idea
23:01.54Maxxed26XX
23:02.32dlynesMaxxed: ${EXTEN:2} means strip the first two digits from the extension, and return the result
23:02.44Maxxedright right
23:02.57Maxxedoh!
23:02.58Maxxeddamnit!
23:02.59dlynesMaxxed: So ${6041234567:2} equates to 41234567
23:02.59Maxxedhah ;p
23:03.49Maxxederum.. so im backwards
23:03.53*** join/#asterisk saftsack (n=saftsack@p54A7D6F7.dip.t-dialin.net)
23:04.00Maxxedwell, how would i flip that?
23:04.02dlynesMaxxed: apparently :)
23:04.08Maxxedhar har har ;)
23:04.24bkw__V A D
23:04.26bkw__woohoooooo
23:04.56Maxxed26XX.. er..um.. uh..
23:05.38*** part/#asterisk swytch (n=ezcall@d83-177-197-138.cust.tele2.fr)
23:05.56Maxxedso um, like.. uh
23:06.00Maxxedim still drawing blanks
23:06.07Maxxedi know this is stupid easy
23:06.12dlynesSubstrings
23:06.12dlynes<PROTECTED>
23:06.19dlynesStraight from the variables wiki for asterisk
23:07.00Maxxedi still dont follow :\
23:07.14Maxxedhaha ;)
23:07.23dlynes${EXTEN:0:2}
23:07.31Maxxedbeh..
23:08.05MaxxedGotoIf($[${CALLERIDNUM} = {$EXTEN:0:2}]?3:5)
23:08.14dlyneseh?
23:08.18Maxxeddont i need some " " 's in there or like
23:08.28dlynesthe '$' goes before the '{', not after it
23:08.33Maxxedwoops
23:08.33Maxxedsry
23:08.37Maxxedtype o
23:08.49dlynesAnd it's ${CALLERID(num)} in 1.2 and higher
23:08.57Maxxedah
23:09.09Maxxedi havent done anything with asterisk hardcore in a good long while
23:09.19*** join/#asterisk Greek-Boy (n=fusion@193.220.93.162)
23:09.37dlynesI haven't seen asterisk on any hardcore sites at all
23:09.46*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
23:09.47Maxxedhar har har :p
23:09.48*** join/#asterisk darkgumby (n=jfox@d1c47c0e.gw104.dsl.airmail.net)
23:09.48Maxxedheh
23:09.56Maxxedwell it is the hottness
23:10.00Greek-Boyhi dlynes :)
23:10.06dlynesHeya Greek
23:10.23Maxxedso GotoIf($[${CALLERIDNUM} = ${EXTEN:0:2}]?3:5) should work?
23:10.25dlynesYeah...personally I thought tintin was much sexier than his dog ;)
23:10.34Maxxedit looks as if im missin' somethin'
23:10.49dlynes2622 would become 26
23:10.59dlynesIs that what you want?
23:11.11Maxxedkinda sorta
23:11.30dlynesWell, I would assume the only two possible answers would be yes or no
23:11.42dlynesIt's an either or question :)
23:11.48Maxxedany caller that has an ext between 2600 and 2699 will go to 3
23:12.27darkgumbyNewbie here needs some help with A@H 2.8 - Am I in the right place?
23:12.46*** join/#asterisk g4m (n=graham@dsl253-014-003.sba1.dsl.speakeasy.net)
23:12.52dlynesGotoIf($[${CALLERID(num):2:2} = 26]?3:5)
23:13.02LostFrogdarkgumby: #FreePBX
23:13.03Maxxedah hah!
23:13.09Maxxeddamnit, i can see it, but i cant think it
23:13.21darkgumbythnks LostFrog
23:14.08dlynesWhy do sparcs have to have 25 pin rs232a connectors?
23:14.12LostFrogThat looks backwards, dlynes, shouldn't it be :0:2?
23:14.31dlynesLostFrog: Yeah...actually..
23:14.38dlynesLostFrog: heh...thanks for spotting my mistake
23:14.54g4mDoes anyone have a recommendation on a good way to present a list of what SIP clients are in use? I want to create a webpage or some way to access the information presented with "sip show channels" from the asterisk command line tool.
23:15.00dlynesMaxxed: See LostFrog's correction to my solution
23:15.34*** join/#asterisk naturalblue (n=Administ@87.192.100.109)
23:15.35dlynesg4m: You mean what hard phone/soft phone is connected and which version it is?
23:16.00Maxxeddlynes: i got it
23:16.16Maxxeddlynes: i wanted to test it ve4 i said anything ;p
23:16.17g4mdlyens: no, i want to figure out who is currently making a call.
23:16.20dlynesg4m: sip show channels would show which sip clients are currently involved in conversations, not which sip clients are in use
23:16.23Maxxedthanks a bunch dlynes :)
23:16.32*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.116.Dial1.SanJose1.Level3.net)
23:16.42g4mdlynes: i mispoke, i'm trying to figure out what clients are having conversations.
23:16.56dlynesg4m: so exactly what you said then...sip show channels
23:17.19dlynesg4m: assuming you only care about sip clients
23:17.32dlynesg4m: show channels if you're concerned with all
23:17.56g4mdlynes: Is there a way i can get that information via AGI or some other non-cmdline tool.
23:18.19dlynesg4m: well, you could always do asterisk -rx 'sip show channels'
23:18.31dlynesg4m: from a cgi script
23:19.06Zodiacalanyone know if callerid on analog lines allways has a delayed ring? is there a way for it to ring instantly on both sides but just have the callerid info delayed?
23:19.19dlynesg4m: agi might be able to do it, too...I just haven't tried using agi yet, so I'm not entirely sure what it's capable of
23:19.20LostFrogI'm sure if you do it in perl you could parse it fairly easily.
23:19.36dlynesg4m: You can also do it, using the asterisk management api, too
23:19.56g4mdlynes: cool, i'm trying to avoid using anything that requires raising privs.
23:19.58dlynesg4m: but of course using that method, if you do something bad, you can fubar the server
23:20.26dlynesg4m: nobody said you had to run asterisk as the root user
23:20.40g4mdlynes: true.
23:21.25dlynesg4m: but yeah, probably what you're looking for is the asterisk manager
23:21.38hydridetheres no debian package for freepbx is there?
23:21.53dlyneshydride: you could try asking on #freepbx to find out
23:21.59hydridethanks
23:22.21dlynesg4m: asterisk manager by default listens on port 5038
23:22.42copantlcan i use diferents contexts in oh323?
23:24.57*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
23:25.27*** part/#asterisk terrapen (n=cjs@166.70.183.108)
23:33.13*** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
23:40.57De_MonI read about a app_cbmysql that support postgres and mysql, the version that comes with web-meetme 2.0 doesn't support it though
23:41.40De_Monand what is this MeetMe2 I keep reading about?
23:49.48*** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz)
23:50.35*** join/#asterisk zeppelin_ (n=zeppelin@201.41.1.202)
23:53.18dlynesDe_Mon: Have you tried res_odbc?
23:53.32*** join/#asterisk Lizard-Ki (i=Hcard@grimlock.macroshell.com)
23:54.46*** join/#asterisk Maxxed (n=whyman@65.59.245.122)
23:54.50Maxxedha hah! back again!
23:55.13Maxxedim nooow trying to intercept dtmf tones while a call is in session
23:55.26De_Monno... not even sure what it is
23:55.30Maxxedi saw a voipwiki page that covered it some
23:55.34Maxxedbut hell if i cant find it
23:56.48dlynesDe_Mon: it's the recommended database driver for asterisk; it's kept current...the other ones aren't
23:57.29De_Mondlynes cbmysql looks like it's doing some magic to interface with the meetme web interface
23:57.50De_Monit's all very badly documented :|
23:58.11dlynesDe_Mon: no idea what meetme web interface is
23:58.38*** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
23:59.20*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
23:59.22De_Monhttp://www.voip-info.org/wiki/view/MeetMe-Web-Control
23:59.47De_Mondlynes I've been trying to wrap my head around it for the last 2 days and it's still pretty vague.

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