IRC log for #asterisk on 20080119

00:00.48jblackOhhh, wait. Enum and dundi changes
00:00.54russellb:)
00:01.01russellbDUNDIQUERY and ENUMQUERY are hot
00:01.08russellbi wrote DUNDIQUERY, hehe
00:01.15russellbENUMQUERY was done by my younger brother ..
00:01.25jblackYay! No more ugly agi scripts in order to hunt in priority order?
00:01.34russellbheh
00:01.54russellbi suppose so, if you're talking about processing multiple ENUM or DUNDi results
00:02.28jblackYup. rather than contemporary ENUMLOOKUP provies one randomized answer, or somesuch.
00:03.06russellbyeah ... and if you wanted to look at the other results, you do the ENUM lookup multiple times
00:03.11russellband you might not get the results in the same order ..
00:03.12russellbit was ugly
00:03.24WilliamKrussellb, so when does faxing get fixed?
00:03.26jblackPlease, please tell me that there's a new foreach?
00:03.39russellbforeach?
00:03.42russellbfor what, enum/dundi?
00:04.29russellbWilliamK: "fixed" is pretty broad ... but i can assure you that it is being actively worked on
00:04.32russellbjblack: http://www.asterisk.org/node/48356
00:04.37russellbthat's a usage example
00:04.40russellbprobably what you mean ...
00:05.29russellb1.4 backport is available ... svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4/
00:06.10russellbENUMQUERY has the same interface
00:06.13jblackThis is completely made up out of my mind and bears no relationship to anyone's reality, but : http://pastebin.com/m4fb7e493
00:06.16russellbor, close enough ...
00:06.19WilliamKrussellb, a nice word is to have something that actually works
00:06.34russellbjblack: yeah, that's basically how it works
00:06.34WilliamKit's been a major sticking point
00:07.19jblackOh! While loops????
00:07.27WilliamKand this google search thing didn't help digium much either, I found it pretty appalling from the view of a system integration/hosting standpoint
00:07.38jblackI don't remember reading that in the patchlog. Has that been in 1.4 all along?
00:07.46russellbjblack: while loops?  yeah
00:07.52russellbnot the QUERY stuff
00:08.40jblackDo you know how many fugly goto loops I've set up?
00:10.01nhuisman_work*cross his fingers for abe v C*
00:10.10russellbheh, thanks
00:11.44russellbWilliamK: see Danny Windham's post to -biz
00:12.08Qwellrussellb: has that already been sent? O.o
00:12.34Qwelloh, guess it has
00:12.59fileso it has!
00:13.05Qwellhttp://lists.digium.com/pipermail/asterisk-biz/2008-January/025050.html
00:13.07Qwellfor reference
00:13.35Qwellit still takes a while for list email to get to me, for some reason..
00:14.06lmadsenI just saw it in my inbox
00:14.21QwellI often don't get list email for an hour after it's sent...
00:14.26lmadsen-biz and -users never gets to me :)
00:14.28Qwellincluding svn, which is pretty annoying
00:19.10drmessanocontemplates too
00:19.17Qwelldrmessano: it's high volume
00:19.30drmessano# per day?
00:19.43Qwellsometimes 50+?
00:19.49Qwelllet's see
00:19.53drmessanoEh.. I get that on most of my ham radio lists
00:19.57drmessanoSo thats ine
00:20.11Qwellyeah, 50+ some days
00:20.18drmessanoThats tolerable
00:20.46drmessanoOf course, I my IRC trolling takes a lot of my time
00:20.56drmessanoI may not be able to participate properly :(
00:21.02drmessanogod man
00:21.07Qwelljust s/trixbox forums/asterisk-users list/
00:21.11drmessanolol
00:21.16Qwellplenty of free time
00:21.40drmessanoActually, I can speed read the forums now
00:21.53drmessano_ is broken in 2.4
00:21.57drmessanoZOMG ADS
00:22.06drmessano___ help pls Zaptel
00:22.36drmessanoYou guys are the best ever (insert sucking upage)
00:22.44drmessanosort, rinse, repeat
00:23.29*** join/#asterisk wyoming (n=steve_mu@216.166.159.235)
00:23.46drmessanoChris Lymans rantings are much more enjoyable :)
00:24.10tzangerlmadsen: you need to subscribe, idjit
00:24.41lmadsentzanger: this assumes I *want* to read it :)
00:24.45tzangerooh that would be nice
00:24.52tzangersubscribe him to LKML while you're at it
00:25.31drmessanoOh, and add him to your facebook
00:25.55lmadsenprobably already done for a bunch of people :)
00:26.05tzangerI have never been on facebook
00:26.14drmessanolol
00:26.22Qwelllmadsen wiishes he could be my friend on facebook
00:26.31drmessanoFacebook is cool.. Its the MySpace you dont have to be ashamed of
00:27.00russellbit's like myspace, except it isn't absolutely terrible
00:27.13drmessanoMySpace: HALO R U O K?
00:27.23drmessanoFacebook:  HALO R U O K? Finish that proposal?
00:27.38tzangerheh
00:27.47tzangerhttp://youtube.com/watch?v=1Q9qSirurmU&feature=related
00:27.49tzangermore heh
00:27.57russellbQwell: signage
00:28.12russellbponders?
00:28.26Qwellat home, not sure I have key here...  BUT, vpn
00:28.28tzangerAYPWIP?
00:28.33jblackI just realized that having * running properly can literally be a matter of life or death.
00:29.14drmessanoROFL
00:29.38drmessanojblack: Asterisk is not a game.. but games may be played with it
00:29.48jblackSeriously. Put * into an office and set up a bad dialplan that renders 911 nonfunctioning. Now nobody can call 911 while someone's having a heart attack. Or a stroke, or chokes to death.
00:30.12drmessanoThats not a big deal overseas
00:30.16tzangerjblack: yes, and there aren't such things as cell phones
00:30.24tzangerbut you're right, this stuff needs to be tested
00:30.37tzangeralso verify with the PSAP that your CID is correct to them and has the right address registered
00:31.15Qwellrussellb: done!
00:31.44jblackI'm not trying to be melodramatic or anything. Just musing about a risk that doesn't really exist with postfix or apache.
00:32.09tzangerah
00:32.25drmessanoYes, which is why idiots can be dangerous
00:32.33drmessanoor people who get in over their heads
00:33.10drmessanoI R MAKE PEEEBEEEX NOW <-- one side effect of having an open source telephony engine
00:33.41jblackAbsolutely true. Idiots have all sorts of ways to be dangerous to themselves and those around them.
00:33.58*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-76-246-191-68.dsl.lsan03.sbcglobal.net)
00:34.05jblackThey can outright kill someone with a car, or a strategically rammed pencil. :)
00:34.16*** part/#asterisk jpeeler (n=jpeeler@216.207.245.1)
00:35.29drmessanoYep.. and they have the uncanny ability to get jobs being IT admin and fry cooker
00:35.49jblackdon't worry about him. linuxmce flummoxes him
00:36.35drmessanoSo does the weight of the concrete bottom of a tennis court
00:36.39drmessanoRIP MrDigital
00:37.45drmessanoNo.. but you can ask them for a reduced rate line for incoming only, and 911 out
00:37.46lmadsenjblack: I tell most people not to rely on VoIP for anything, especially phone calls :)
00:37.59jblackvirtually over ip, eh?
00:38.49drmessanoVariably Operating Internet Phone?
00:39.06lmadsen:D
00:39.16lmadsenVariably Operational Internet Phone
00:39.29x86efnethaha.
00:39.44lmadsenmy side burns need trimming...
00:39.57drmessanox86efnet
00:40.04drmessanoYoure not on efnet
00:40.08drmessanoSorry :(
00:40.18x86efnetI'm from there.
00:40.41lmadsenefnet's been around for a pretty long time...
00:40.42drmessanoSad
00:40.47drmessanoThats.... sad
00:41.04drmessanoWhen youve been online long enough that you claim it as your hometown
00:41.08drmessanoIts time to go outside
00:41.08TJNIIWhat's the CLI command to stop music on hold on a channel?
00:41.19x86efnetlol..
00:41.54drmessanox86efnet: As a 17th level Madrook of the High Order of Phanpangel, I order you to go outside
00:42.40x86efnetahhhhhhhh...... are you really level 17 dratz!
00:43.12drmessanoI would be at level 18, had not the day rune
00:43.38jblackI'm a level 43 dungeon master.
00:43.53drmessanoI believe it
00:43.53Davieyjblack is my hero
00:43.55drmessanoReally
00:43.58drmessanoI believe it
00:44.22*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
00:44.47jblackYou kids really missed out on D&D.
00:45.05drmessanoYou missed out on sunlight and kissing girls
00:45.30Qwellsunlight?  who needs that?
00:45.30jblackWhat with the computers and all.. When I was a kid, a bunch of nerds would get together to play a game. We'd spend 4 hours preparing, and 20 minutes actually playing.
00:45.39jblackWhat is this sun thing you speak of?
00:45.47drmessanolol
00:45.53Qwelljblack: 20 minutes?  not 18 hours?
00:45.54Qwellwussy
00:46.13Qwellor 30 hours on those really hardcore days :p
00:46.22jblackNot after the 4 hours of screaming.
00:46.23drmessanoMoving from NJ to GA when I was a teen was the best thing that ever happened to me.. I could stand the heat, so I spent all my time indoors on the computer lol
00:46.46drmessanocouldnt*
00:46.50DavieyI've been in a D&D battle for the last 27 years
00:46.55drmessanoROFL
00:47.03Davieywe are hardcore
00:47.03jblack"There is no way you rolled 6 18s."...
00:47.14Qwelljblack: heh
00:47.36*** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta1 (2008/01/18), Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org)
00:48.17jblack"Did too! I spent 8 hours randomly rolling until I got that." "Liar!" "Well, where exactly did you find that +45 bastard sword?" "YOUR BOTH LIARS!"
00:48.18lmadsenrussellb: congrats!
00:48.26jblackAnd that's before the first round of cokes are down.
00:48.35lmadsenjblack: you're both liars :)
00:48.54drmessano"I want to cast a spell"
00:48.59drmessano"Magic Missile"
00:49.14drmessano"What do you want to cast magic missile that, there's nothing here?"
00:49.23drmessano"I want to cast it at.... the darkness"
00:49.33lmadsengood band
00:49.33jblack"With an intelligence of 16, that'll take you 2 rounds. An orc  runs up and kicks you in the balls, causing you to miscast, thereby shooting yourself in the balls too"
00:49.42x86efnetD&D kicked ass, you guys ever play the old door game lord?
00:50.06lmadsenx86efnet: you must be young :)
00:50.11jblackWho didn't play lord?
00:50.17jblackI like empire, myself.
00:50.20lmadsenBRE!
00:50.28jblackYeah, barron realms.
00:50.31x86efnetyeah BRE and empire where cool.
00:50.39x86efnetI'm 23
00:50.45lmadsenI like Usurper
00:50.49jblackI never could get enjoyment out of tradewars, though.
00:51.06x86efnetsame here jblack,
00:51.18jblackdrmessano: Telephony pr0n?
00:51.21drmessanolol
00:51.38x86efnetATTH0+++
00:51.38AdamWestascii pr0n was the best
00:51.54AdamWest(.  )(.  )
00:51.56jblackHey, do you think I could get digium to delegate pr0n.asterisk.org to me, so that i could put up a telephony pr0n site?
00:52.36drmessano"jblack slowly slide the cover off the box, exposing the naked PCI slots beneath"
00:52.36*** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net)
00:52.37AdamWestsorry, that's the backup business model if hardware doesn't work out
00:52.41drmessanoslid*
00:52.44x86efnetthat'd be pretty retro dude .. what about ascii.asterisk.org so you could have it in sub catogorys.
00:52.58jblackDamn.
00:53.15jblackJust imagine all the compromising positions we could put a slom and a polycom into....
00:53.17*** mode/#asterisk [+o codefreeze] by ChanServ
00:53.26drmessano"jblack shuddered as he recompiled his zaptel"
00:53.58AdamWestso *this* is what asterisk people do on a friday night... I see
00:54.02jblackI have this imagary stuck in my head, now, of someone throwing a bucket of water over a 66block.
00:54.03drmessanolol
00:54.11jblackA'la flashdance
00:54.20drmessano"My god, I don't think i've ever seen a kernel that big before"
00:54.32*** join/#asterisk BBHoss_Laptop (n=bbhoss@76.73.251.16)
00:54.46jblackIs that handset in your pocket, or are you just glad to voip me. ;)
00:54.54BBHoss_Laptophey anybody know anybdy that sells Laos DIDs?
00:55.13jblackyeah.. some people go to the bar, get a chick drunk enough to get her home, and come up with all sorts of interesting things to do.
00:55.24drmessano"She slowly peer'ed out the window, wishing for one last SIP of the IAX she had longed to trunk"
00:55.26jblackMe? I want to put an SPA-8000 in a teddy.
00:55.28DavieyD&D!
00:55.34*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
00:55.40BBHoss_Laptopdrmessano lol
00:55.51*** join/#asterisk kamanashisroy (n=root@202.56.7.141)
00:55.57jblackOHH! I know what we can call the site!!!
00:55.57*** part/#asterisk kamanashisroy (n=root@202.56.7.141)
00:56.00jblack"Phone Sex"!
00:56.11*** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net)
00:56.35Davieyfor those on the mailing list, "She could asSense the hostile feeling towards being branded"
00:56.37drmessanoIt needs to be more risque... implying danger..and excitement...
00:56.38drmessanoI KNOW
00:56.40*** join/#asterisk wyoming (n=steve_mu@216.166.159.235)
00:56.43drmessanoWe can call it SVN
00:56.44drmessanoWait, no
00:57.11jblackNOTE TO SELF: Either tell less funny jokes, or quit smoking.
00:58.46drmessano"She waited as he booted.. feeling the warm buzzing of his fans... Just as she couldnt take the excitement anymore, disappointment set in as she discovered her worth fears were true..  He was a Trixbox"
00:59.17drmessanoworst* crap
00:59.39jblack"Hey, big boy. Plug your rj11 into my socket any time"
01:00.15jblackOhhhhh. You're so biiig.. you must be an rj45
01:01.00drmessanolol
01:01.04x86efnetI love when cepstral says that in the callie voice
01:01.40drmessanoI love how you can chain words together and make Allison say almost anything you want.......  not that ive done that
01:02.16Qwellor, you could hire her to say it...professionally
01:02.27*** join/#asterisk newave (n=newave@74-135-100-180.dhcp.insightbb.com)
01:03.39drmessanoHmmm.. i'm sure she morals that I would be WAY to tempted to test the boundaries of
01:03.43drmessanoshe has
01:03.44drmessanoWow
01:03.51drmessanoMy typing is AWFUL
01:03.58Qwellthey've been tested
01:04.03drmessanolol
01:04.07drmessanoGood to know
01:04.07QwellI know of one word that she won't say.
01:04.19drmessanojackmuffin?
01:04.31Qwellmaybe it was because of who asked for the prompts though...dunno
01:04.36drmessanolol
01:05.11jblackAllison seems to have a great sense of humor. Do you think she'd be ok with the innuendo?
01:05.56riddleboxQwell, does it start with a C
01:05.58jblackOh, did anyone see that movie?
01:06.04jblackCloverfield?
01:06.15riddleboxI want to see it
01:06.26newavedoes a 'sabre' chill in here by chance anyone?
01:06.29jblackMe too. It came out today.
01:06.40Qwellnewave: doesn't sound familiar
01:06.42Qwell~seen sabre
01:06.59jbotsabre <~sabe@pcp462384pcs.lvylok01.ar.comcast.net> was last seen on IRC in channel #asterisk, 1131d 22h 54m 29s ago, saying: 'does anyone know where i can find docs on loadbalancing multiple servers with asterisk?'.
01:06.59newavehe may have changed his nick
01:07.05riddleboxmy girlfriends godson made us take him to see Alien Vs Predator Requilm, wow it sucked
01:07.39newave1131 days! wow
01:07.46drmessanolol
01:07.50jblackjbot: pr0n.asterisk.org is reserved by the corporate overlords as a backup plan.
01:07.51jbotjblack: okay
01:08.01*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)
01:08.17riddleboxlol
01:09.26drmessanoWhat would be a good linux app for a customer initiated remote session?
01:09.34jblackFor doing what?
01:09.40drmessanoWell
01:10.07jblackHow about... MeetMe
01:10.33drmessanoFor VNC they have the PC HelpWare app.. which creates an outbound remote desktop session to a proxy, which, I, the support person, can attach to by connecting to the same proxy and authing
01:10.45jblackOhhh.
01:10.53jblackrdesktop
01:11.07jblackIt's free software and uses microsoft's protocol.
01:11.15jblackI thik it's free software
01:11.35drmessanoYeah, but does it initiate the outbound session?
01:11.38jblackYeah. It's free software.
01:11.40drmessanoThats the key
01:11.49jblackYou can rdesktop to things, if that's what you mean.
01:11.52drmessanoNo
01:12.05*** join/#asterisk AndyGraybeal (n=andy@node230.39.251.72.1dial.com)
01:12.12drmessanoThe customer starts the APP
01:12.16newaveWell I dont see who I am seeking, so thank you, have a good one everyone!
01:12.27jblackAnd it magically appears on your desktop?
01:12.31drmessanoDesktop shared out --------------------------> Proxy on stream 12
01:12.40drmessanoI connect -----------------------> Proxy at stream 12
01:12.54*** part/#asterisk newave (n=newave@74-135-100-180.dhcp.insightbb.com)
01:12.59drmessanoCustomer -------------------------> Proxy <-------------------------- Me
01:13.09drmessanoNAT transversal is handled by the proxy
01:13.28*** join/#asterisk TokyoJimu (n=TokyoJim@adsl-64-168-153-93.dsl.snfc21.pacbell.net)
01:13.29drmessanoActually the proxy handles everything
01:13.35jblackI'd check to see if rdesktop and vnc can do socks
01:13.44*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)
01:14.03drmessanoIts not just that
01:14.05tzafrir_homedrmessano, why not use ssh
01:14.06tzafrir_home?
01:14.18tzafrir_homeand tunnel whatever you want on top of it? e.g: VNC
01:14.20TokyoJimuHi.  Running cdr_mysql on * 1,2, if it can't contact the DB it stops taking calls and all callers get reorder.  Any way around this???
01:14.28drmessanoThe idea would be the customer would initiate outbound
01:14.52jblackHow about....
01:14.54drmessanoSo no need for firewall penetration, NAT transversal, etc
01:14.58*** join/#asterisk fnordus (n=dnall@24.84.160.227)
01:15.09jblackSetting them up with putty to do port forwarding to your machine.
01:15.12tzafrir_homeVNC also allows the server to connect to a client
01:15.25tzafrir_homeRTFM
01:16.04drmessanoI need to look into that then
01:16.15drmessanoIm assuming RealVNC?
01:16.22drmessanoTheres only 100 flavors
01:16.58jblackwell, great excuse to make #101
01:17.01*** join/#asterisk xlotlu (n=john@79.114.174.42)
01:17.13drmessanoI know UltraVNC will "Add Clients"
01:17.21drmessanoAnd it can connect to the VNC repeater
01:17.35drmessanoDidnt know anyone else had implemented that, let along on Linux
01:17.38drmessanoalone*
01:17.56*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
01:18.15drmessanoIt would be nice to put on a clients PBX for remote access
01:18.25drmessanoNot having to deal with firewalls
01:18.36drmessanooutbound, yes
01:18.43drmessanoBut thats not usually a problem
01:18.43jblackit seems to me that letting them pop their desktop up on your automatically will make it harder for you to avoid them.
01:18.53drmessanoNo
01:18.57drmessanoIts doesnt work that way
01:19.13drmessanoThey initiate to the proxy (repeater) or client..
01:19.16tzanger[TK]D-Fender: I got that contract in Montreal
01:19.21drmessanoIn this case, there would be a waiting proxy
01:19.26drmessanoand I would need to attack to it
01:19.30drmessanoattach*
01:20.21jblackI get it. You want a desktop router that is akin to asterisk.
01:20.35drmessanoWell, its there for Windows
01:20.44drmessanoThe Repeater exists, as does the other parts on Windows
01:20.46drmessanoIve used it
01:21.03drmessanoI can send someone an EXE, tell them to open it, connect to the repeater, and bam I am in
01:21.13drmessanoNo open ports, exchanging IPs, nothing
01:21.35drmessanoBut didnt know if there was either a VNC implementation on Linux that supported it, or another solution
01:24.53drmessanowow
01:25.02drmessanoRealVNC has REALLY gone commerical
01:25.09drmessanoNew website, etc
01:25.15*** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com)
01:25.21kronchthey were started by att.
01:25.34drmessanoYeah
01:26.00nhuisman_workpattern matching is used for outgoing calls right, can this work for internal numbers too, so I don't really have to put any of my hundred phones in the extensions.conf
01:26.12drmessanoIt went from Free  -----> We're charging.. please buy ------- Trixbox
01:26.13drmessanoErr
01:26.13nhuisman_worktrying to learn this stuff before I start created an extensions.conf
01:26.17drmessanoI didnt say that
01:27.20*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
01:28.44*** join/#asterisk pdu (n=kvirc@81-86-38-170.dsl.pipex.com)
01:29.12*** join/#asterisk adjohn (n=adjohn@p1089-ipad76marunouchi.tokyo.ocn.ne.jp)
01:31.50[TK]D-Fendertzanger, Cool... you were working off Mississauga prior, right?
01:32.15*** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca)
01:34.59*** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)
01:37.46pduevening all
01:38.12jblackWTF With the morons these days?
01:38.19tzanger[TK]D-Fender: markham, yeah
01:38.51jblackMy online broker came up with a "brilliant idea".... To log in, instead of typing a username and password, they're going to require flash9 so that you can enter your password by clicking on a graphic.
01:39.02[TK]D-Fendertzanger, wasn't it off Pearson though?
01:39.37tzanger[TK]D-Fender: no
01:41.03[TK]D-Fendertzanger, Hrm... not sure who I mixed that up with... oh well.  So you moving this way?
01:41.05pdui've been doing some googling and found a few articles talking about an intel winmodem being the same as an X100 fxo module
01:41.14drmessanolol
01:41.18pdubut found nothing about any other modems, does anyone have any experience with this?
01:41.36[TK]D-Fenderpdu, Yes, the Inten 537 chipset = X100
01:41.39*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
01:41.39*** mode/#asterisk [+o anthm] by ChanServ
01:41.44drmessano~x100p
01:41.45jbotx100p is, like, an obsolete card.  You don't want to bother trying to make it (or any of the "digium compatible" clones) work.  Get a TDM01B, and you will save your sanity, your hair, and countless other things.
01:41.48[TK]D-Fenderpdu, And other modems = unusable.
01:41.58[TK]D-Fenderpdu, and as it is, the X100P was crap
01:41.59tzanger[TK]D-Fender: no, just a contract, I'm working in Verdun area
01:42.03tzangerer I will be
01:42.05pdulol ok thanks
01:42.22[TK]D-Fendertzanger, Cool, we'll have to grab a beer sometime
01:42.31pduall my questions answered lol
01:45.53*** join/#asterisk denon (n=denon@tooth.decay.org)
01:45.53*** mode/#asterisk [+o denon] by ChanServ
01:54.24NivexI don't see chan_mobile in the 1.6.0-beta1 release.  Is it safe to assume it got dropped?
01:54.34nhuisman_workwhat's chan_mobile?
01:54.37nhuisman_workmobile phones?
01:54.56Nivexyeah.  Interface to consumer mobiles over bluetooth.
01:55.03Nivexuse your cell phone as an FXO port
01:55.08nhuisman_workdamn
01:55.11nhuisman_worki was interested in that
01:55.17NivexI saw it in SVN awhile back, but admittedly I have been lax in following lately
01:55.28NivexI rejoined here when I saw the beta1 release
01:57.37nhuisman_workthat's alot of changes
01:57.55ZX81_this work for anyone else? http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.6.0-beta1.tar.gz
01:58.04ZX81_just takes me to empty directory
01:58.35Nivexoh, nm, it's in asterisk-addons
01:58.59tzafrir_homehttp://downloads.digium.com/pub/asterisk/releases/asterisk-1.6.0-beta1.tar.gz
01:59.02nhuisman_workZX81_, yes it's borken
01:59.08ZX81_ok cool
01:59.22nhuisman_worktzafrir_home, so is that link
01:59.39ZX81_yeah so another press release with no download lol - all good :)
02:00.08tzafrir_homegrab it from the SVN . We don't need no tarballs :-)
02:00.58ZX81_just for the adn
02:00.59ZX81_~adn
02:01.00jbotmethinks adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php  for HTML and http://feeds.feedburner.com/asterisknews for RSS
02:01.12ZX81_I've put up the link that is on www.asterisk.org but its down
02:01.41*** join/#asterisk adjohn (n=adjohn@p1089-ipad76marunouchi.tokyo.ocn.ne.jp)
02:04.07*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
02:11.19x86efnet0xb7da21b8 <CAST_S_table0+61432>:        "strlen(objstr)+23+2*enc->iv_len+13 <= sizeof buf"
02:11.27x86efnetpossible security issue found there.
02:12.14ZX81_open an entry on the bugtracker
02:12.18ZX81_bugs.digium.com
02:12.41ZX81_maybe mention it in #asterisk-dev
02:20.04drmessanoAIM = XMPP now
02:20.21drmessanoI need to play with the Asterisk config a little and see if I can connect
02:20.46*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
02:27.45*** join/#asterisk d-k-t (n=dt@60.176.192.94)
02:31.34drmessanoBAH
02:31.39drmessanocant get a user connected
02:32.42*** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.ca.comcast.net)
02:33.35OlobolaCan ael2 run off a seperate server the same way fastagi can?
02:35.15hmmhesaysI wish there was a way to playback a sound to both channels with a dynamic feature map
02:49.17jblackDidn't [TK]D-Fender tell you how to do it about 4 hours ago?
02:54.15drmessano5 hours 39 minutes ago
02:54.46drmessanoBut who is counting?
02:55.28jblackheh
02:55.53jblackI think I see how one could do it with Dynamic_Features.
02:57.42jblackYeah... Huh.
02:58.25hmmhesaysheh, good luck gettting the proper variables to create the call file
02:58.28hmmhesaysbecause I was thinking the same thing
02:58.43*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
02:58.44jblackI imagine one could use AGI as an application, create a new channel, have it chanspy on the other two channels, and whisper to both of them.
02:58.52jblackI bet there's an even simpler way.
03:00.02hmmhesayshow are you going to call it?
03:00.19hmmhesayshow are you going to tell chanspy which channels to spy one
03:00.19jblackthe agi?
03:00.20hmmhesays*on
03:00.43jblackI imagine that there's one or two variables that list the channels that the feature came from.
03:00.51hmmhesaysdynamic features execute outside of the core, you don't have access to the normal channel variables
03:01.44jblackDid you check the environment?
03:02.04jblacksetup a dynamic_feature to point to an agi, and get it to rattle off what it _does_ know.
03:02.45jblackIf you can get to an agi while keeping track of the channel, then you can take over the world.
03:03.09jblack*(note: taking over the world requires accessories such as nuclear tipped missiles, large standing armies, and sharks with laser beams on their heads)
03:03.42*** join/#asterisk postconf (n=marquis@gw-corp.postconf.com)
03:03.59jblacksure, you're not going to have context variables, but there may be something in there that gives the channels.
03:07.05jblackFailing _that_, write an asterisk application module that you can call.
03:07.56jblackdrmessano: So, I guess we've got another 5 hours before he comes back again?
03:07.59*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
03:08.40*** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo)
03:08.43drmessanoPete and Repeat were in a boat
03:08.55drmessanoPete fell out
03:08.58drmessanolol
03:09.10jblackThe first line alone is worth a smile.
03:10.18drmessanoI feel like a total jerk
03:10.30jblackwhy?
03:10.44*** join/#asterisk ZX81 (n=ZX81@202.20.97.211)
03:10.45drmessanoTrying to get the jabber integration connecting to AIM
03:10.51drmessanoI feel like.. a sellout
03:11.00jblackAren't you a little far ahead of the curve?
03:11.18jblackDidn't aol just announce their intentions today, and just starting work on that?
03:11.25drmessanoNo, its up
03:11.37drmessanoBeta.. but its up
03:11.44drmessanoPeople are using it
03:12.34*** part/#asterisk supjigator (n=shanebur@152.53.16.10)
03:13.22jblackeither that, or an army of starving children, that can live in his basement with a slice of bread for daily food, that can perform dtmf tones on behalf of callers.
03:13.32drmessanoHA
03:13.42drmessanoJust 30 cents a day
03:13.44jblackIt works for shoes!
03:13.45*** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net)
03:17.56*** join/#asterisk _ShrikE-cell (n=_ShrikE-@32.129.143.117)
03:19.20*** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net)
03:19.27jblackHardy Heron isn't.
03:24.20*** part/#asterisk postconf (n=marquis@gw-corp.postconf.com)
03:33.13*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
03:33.13*** mode/#asterisk [+o russellb] by ChanServ
03:40.53*** part/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo)
03:43.20*** join/#asterisk postconf (n=marquis@gw-corp.postconf.com)
03:45.26*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com)
03:51.08*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
03:56.42*** join/#asterisk cruisefx (n=kvirc@c-69-250-158-97.hsd1.md.comcast.net)
03:57.31cruisefxIf I wanted hardware that would work with Asterisk that would detect answering machines, hangups, silence, and would support DTMF tones while a voice file was playing, what would I need?
03:59.27*** part/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au)
04:00.54*** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au)
04:01.41x86efnetI'm getting this error does anyone have an idea what might be wrong. [Jan 18 22:43:50] WARNING[1335]: res_odbc.c:530 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
04:02.37jblackI'd say you failed to configure res_odbc.conf correctly.
04:02.51*** part/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au)
04:03.06x86efnetyeah, thats what I thought.. it's not my error it's a friend of mine.. I said prehaps something isn't commented
04:05.18*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
04:06.35puzzledhi
04:06.40jblackhi
04:07.15puzzledanyone know a free service that I can use to test a 1-800 number from outside of the US. can't get FWD to work
04:08.06puzzledjblack: the FWD time extension worked for me but no 1-800
04:08.19jblackDid you dial *1800XXXXXXX ?
04:08.22jblackYou need the *
04:08.54puzzledyes I did and also tried with ***1800XXXXXXX as it says on the webpage. does it work for you?
04:09.07jblackGive me a moment, I had to modify my dialplan
04:09.36*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
04:09.58jblackyup. working fine here.
04:10.04jblack<PROTECTED>
04:10.14puzzledhmm strange
04:10.40jblackand fwd is a defined sip friend.
04:10.58puzzledha there's a difference. I have it on iax2
04:11.04puzzledtime to change that I guess
04:11.23jblackYOu got iax2 to work?
04:11.26jblackI never could get that working
04:11.32jblackYou could also set up ENUM.
04:12.00puzzledDial(username:pass@iax2.fwdnet.net/${EXTEN},30) works for me. at least with the time :)
04:12.22jblackTry *${EXTEN}
04:12.50*** join/#asterisk EvilDeshi (n=deshi@75-130-24-153.dhcp.mdsn.wi.charter.com)
04:12.50jblackYou did understand me when I said "The * is necessary"?
04:13.13puzzledyes I did. can you please pastebin the friend section. don't have that one and don't know what it should be
04:13.30jblackI suppose
04:13.48jblackbtw, this is well documented at fwd.
04:14.01puzzledyes off course. let me check that one first
04:14.50puzzledheh that's filled with IAX2 stuff: http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76
04:15.23jblackhttp://pastebin.com/d5144d6fa
04:15.24*** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290)
04:16.00puzzledjblack: thanks
04:16.01jblackYeah. I've seen that a few times.
04:16.26puzzledthat looks quite simple
04:28.52jblackcome google
04:28.56puzzledjblack: fixes :)
04:32.16AdamWestjeebuz, do you people ever go to sleep? :)
04:32.40jblackI took a long nap back in december
04:34.14*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
04:36.07jblackAdamWest: I know who you are.
04:36.23AdamWestI disagree
04:36.33[TK]D-Fendernananananananananananana BATMAN!
04:36.37jblackJust watch your step, or I'll tell everyone else...
04:36.43jblackawww
04:36.51jblackThere goes my grand extortion plot
04:38.49*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
04:38.49*** mode/#asterisk [+o russellb] by ChanServ
04:39.04puzzledjblack: the switch to sip made it work. thanks for your help
04:39.51BurtWardHoly servers, Adam! There's a lot of fixes betwixt 1.4.15 and 1.4.17
04:40.00puzzledtold ya :)
04:40.21BurtWardpuzzled: Just doing my duty as a public citizen.
04:41.13BurtWardChiparrific!
04:41.44jblackHeh. BurtWard is registered.
04:44.22*** join/#asterisk UnixDog (n=unixdog@adsl-69-234-188-151.dsl.irvnca.pacbell.net)
04:44.56jblackSo... from what I can tell, the stock market worked like this:  GE reported great news, so stocks shot right up. Then George Bush spoke, and the market spent the rest of the day on it's way down
04:45.35jblackAnecdotal proof that some people should just keep their fool mouth shut
04:48.04kyronlol
04:50.26drmessanoI think it would hardcore to commit a bunch of "cosmetic" fixed to Asterisk on April 1st
04:50.46drmessanoI think it would be hardcore to commit a bunch of "cosmetic" fixes to Asterisk on April 1st
04:50.49drmessanoThats better
04:51.33jblackI know what they'll do on April 1st. :)
04:52.14jblack"Commit 88717: Announced partnership with Skype"
04:52.25[TK]D-Fenderjblack, but consider GE is one of the worlds top nuclear weapons developers (We bring good things to DEATH!) once GWB decides to make Tehran glow in the dark GE stock-holders will receive MORE good news!
04:52.40jblackGood point.
04:53.03drmessanochan_skype is now a core module!
04:53.06drmessanolol
04:53.23Nuggetheh
04:53.54jblackchan_iax2 and chan_sip have been deprecated in favor chan_skype
04:54.29jblackwhat is this w/ rhythmbox. Now I can listen to magnatunes and such?
04:54.43cruisefxIf I wanted hardware that would work with Asterisk that would detect answering machines, hangups, silence, and would support DTMF tones while a voice file was playing, what would I need?
04:54.48drmessanoFull support for Paltalk!!! lol
04:55.35jblackcruisefx: An intervention from $DIETY
04:56.01cruisefxjblack: So that's an unreasonable set of requirements?
04:56.19jblackIt's reasonable, but app_amd is broken.
04:56.27drmessano"Digium announces sale of Asterisk to Univision Inc"  ?
04:56.41jblackapp_amd doesn't d the ams
04:56.45drmessano'Asterisk renamed to "Los Telephone De Asterika"
04:56.55drmessano'Asterisk renamed to "Los Telephono De Asterika"
04:57.07[TK]D-Fenderdrmessano, don't forget the new single-drive Zaptel-EveryModem module, removal of all * config files and full port to windows with GUI-only configuration (and a cherry on top!)
04:57.18drmessanoROFL
04:57.28cruisefxjblack: app_amd is answering machine support?
04:57.31drmessanoOh hell yes
04:57.46drmessanoI have a Hayes Smartmodem 9600 I have been WAITING for Asterisk support on
04:57.48drmessanoFINALLY
04:58.48cruisefxjblack: So answering machine support isn't implemented in hardware?
04:59.06cruisefxjblack: Theoretically, any analog card would do?
04:59.10jblackHeh. "NEW: AKERISK mode for console. Includes: O RLY?, app_fixitforme, and func_wutsanat. (Dependancies: Part time work at mcdonalds)
04:59.22drmessanoROFLLLLL!!
04:59.47jblackcruisefx: app_amd actually is supposed to do all the work, but it doesn't.
05:00.28drmessanoapp_tkdfender which checks for abuse of the help command, and if detected, runs rm -rf /
05:00.50jblackdrmessano: The first ten people to svn pull it get free access to the tennis court tunnel. :)
05:01.00drmessanoHA
05:01.06drmessanoRIP MrDigital
05:01.28drmessanoSay Hi to Mr Hoffa for me
05:01.28cruisefxjblack: So I could even do all of that with a lowly voice modem considering that all parts of Asterisk were functioning properly?
05:01.39*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581950.dsl.bell.ca)
05:02.22*** join/#asterisk andrewn (n=andrew@76.191.151.50)
05:03.31*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
05:03.45[TK]D-Fenderdrmessano, the most common form of abuse of "help" : NEGLECT
05:04.28russellbo.O
05:05.14jblackOh, I got another guy I gotta tell you about!
05:05.31jblackBut I can't tell you guys right now.
05:05.40jblackRussel knows who I'm talking about .;)
05:06.58drmessanolol
05:07.13jblackI'll tell you privately. ;)
05:07.50drmessanook
05:11.53jblackoh man.
05:12.17jblackon screen 1 I have about 20 "Private messages from unregistered users are currently blocked due to spam problems,"
05:12.21*** part/#asterisk phr3ak (n=nnnphr3a@22-vh-shost.hostoffice.hu)
05:13.11*** join/#asterisk ManxPower (n=manxpowe@69.2.85.41)
05:19.46*** part/#asterisk postconf (n=marquis@gw-corp.postconf.com)
05:20.49jblackdrmessano: so, that's the story. :)
05:24.53jblackNew rule: There are too many mp3 genres when one of them is "Theater music -improvised alternative"
05:26.49*** join/#asterisk DavidTangye (n=getnikar@30.105.233.220.exetel.com.au)
05:28.15jblackwow. It's the first time in a long while I've heard a pipe organ in rock music
05:28.51[TK]D-Fenderjblack, into to "Let It Rock" - Bon Jovi
05:29.33jblackRight now I'm trying out, and I'm serious here,  "The Commod's"
05:29.47esaymexten => _NXXNXXXXXX/113, is the slash 113 the originating callerid?
05:29.53jblackSo this is what happens when the french sing instead of f... ahh, never mind.
05:29.54[TK]D-Fenderyup
05:29.54*** part/#asterisk DavidTangye (n=getnikar@30.105.233.220.exetel.com.au)
05:30.02esaymnot working
05:30.21[TK]D-Fenderesaym, don't just say it, SHOW IT
05:30.28esaymoh, sorry
05:31.12esaymexten => _NXXNXXXXXX/113,1,Set(CALLERID(number)=2108819677) ;if extension 113 dials out, it needs to change it's callerid number
05:31.42esaymusing that left asterisk saying that it didn't find any matching extentions
05:31.59jblackI've not used /xxx. Let me check the book
05:32.07esaymwhat book you got?
05:32.15esaymI saw that on a website
05:32.22esaymnow I can't find the website :(
05:33.42[TK]D-Fenderesaym, pastebin the failed call attempt at verbose 10, sip/iax debug (etc) enabled, your ENTIRE dialplan context, and your devices config
05:33.45[TK]D-Fender~pb
05:33.51jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
05:36.50jblackI can't find it right now. For some reason, I thought exten => XXXX/#### meant that exten XXXX only exists for extension ####.
05:37.06*** part/#asterisk cruisefx (n=kvirc@c-69-250-158-97.hsd1.md.comcast.net)
05:37.27SwKexten => FOO/BAR,1,whatever
05:37.41SwKfoo is the extension is bar is the caller id num of the incoming call
05:37.41jblackI.E. If I log in with sip and get extension 100, and there's exten => 411/100  , then only I can dial it. The person at exten 101 can't.
05:38.04SwKexample where 5551212 is your ex girlfriends number
05:38.32*** join/#asterisk iamthelostboy (n=nathan@12.187.245.130)
05:38.44jblackoh, ain't this precious.
05:38.58SwKexten =>_X./5551212.1.telezapper()   <-- so any time your ex calls she gets the operator intercept tri-tones
05:38.59jblacktoday's upgrade broke gallery, mjpegtools, nuvexport and smartmontools.
05:39.26jblackOk, so it's what I thought.
05:39.49jblackI didn't pay too much attention to it, as callerid is so easily forged.
05:40.48esaymSwK: yea that's what I though I read, posting pastbin soon
05:40.56jblackHeh. exten =>_X/${EXGIRLFIEND},1,Playback(YouGaveMeHerpesYouCheatingBitch)
05:41.00SwKyeah
05:41.05drmessanolol
05:41.06SwKexactly
05:41.09drmessanoSend(Valtrex)
05:41.09jblackI wonder if allison would record _that_
05:42.20jblack[TK]D-Fender: You could go into the voice recording bus., btw
05:42.43[TK]D-Fenderjblack, o>O
05:43.14SwKyou know there is an Allison TTS engine now
05:43.41iamthelostboyhello :) are usernames between protocols completely seperate?  Could I have a sip user named bob and a iax user named bob?
05:43.47drmessanoI told my wife once that if we ever broke up, I was going to do something horrible to the next guy she met.. give him a positive character reference
05:43.51drmessanoShe wasnt amused
05:44.13jblackiamthelostboy: I believe so. Probably not the best administrative choice you'd ever make
05:44.17SwKiamthelostboy, depends on how you set it up
05:44.25[TK]D-Fenderdrmessano, on the subject of Valtrex : http://ca.youtube.com/watch?v=rHXXTCc-IVg
05:45.28jblackOMG OMG.
05:45.34jblackResults 1 - 10 of about 16 for akerisk. (0.05 seconds)
05:45.35iamthelostboyjblack: why is that, just becomes difficult to manage, trying to figure out who is who? in logs etc, would they not always be prepended with IAX/ or SIP/ ?
05:46.08jblackWell, it can make your callerid logs confusing for one.
05:46.15SwKiamthelostboy, but to answer your question more directly yes you can have sip user named bob and a different iax user name bob... or in trunk (maybe even 1.4) you can have them the same (sorta)
05:46.31jblackespeically if the bob that comes in over sip and the bob that comes in over iax are different bobs
05:46.36drmessanoHAHHAH!
05:46.44SwKwhy use iax anyway
05:47.21SwK(and no iax trunking is not a good excuse)
05:47.24jblackI got chewed out the last way I answered that one, so I'll pass.
05:47.29iamthelostboyright now, i cant really get sip through the network
05:47.30[TK]D-Fenderdrmessano, a personal favourite of mine.... Bill Maher isone of my favourite left-wingers
05:48.08drmessanoI love Bill Maher
05:48.08iamthelostboynot without possibly causing problems for users of that internet connection, and i want to connect in
05:48.14drmessanoThats awesome
05:48.32esaymok folks, her she is: http://pastebin.ca/862485
05:48.39[TK]D-Fenderdrmessano, "Ask your doctor if getting off your ass is right for you!"
05:48.42iamthelostboyso i figure using iax with a softphone should do the trick for me
05:48.43jblackMrs. Akerisk 2007?
05:48.58iamthelostboyplus it just give me another option if i ever need it
05:49.01jblackiamthelostboy: Good luck finding one.
05:49.04SwKwell if you set up SIP correctly it shouldnt affect other users
05:49.21jblackesaym:  What am I supposed to be looking for?
05:49.21drmessanoPete and Pete took a SIP of IAX, now my extensions have been ***ked for weeks, I decided to try it because it was new in trunk, but now my PBX is junk
05:49.23drmessanoword.
05:49.25iamthelostboykiax? iaxcomm?  not too good?
05:49.38jblackAhh, 603 declined
05:49.41[TK]D-Fenderiamthelostboy, SIP <-
05:49.54jblackOk, here you go esaym.
05:49.54SwK503
05:50.02iamthelostboyits the forwarding of a chunk of ports that worries me
05:50.12esaymjblack: esaym, pastebin the failed call attempt at verbose 10, sip/iax debug (etc) enabled, your ENTIRE dialplan context, and your devices config
05:50.13jblackIs that a 5? It looks like a 6 to me. no matter.
05:50.17drmessano1 port or 100000 ports, same listener
05:50.20jblackesaym: Ok, look at line 5.
05:50.28[TK]D-Fenderesaym, Jan 18 23:37:52 WARNING[12843]: pbx.c:2377 __ast_pbx_run: Channel 'SIP/114-007bb640' sent into invalid extension '+12108631936' in context 'default', but no invalid handler
05:50.47esaymyea i see
05:50.54drmessanoThats like saying "If I open TWO ports for apache, its twice as likely to be hacked"
05:50.55[TK]D-Fenderesaym, I don't see * accounting for the "+" you felt necessary to shove in front of the number you dialed...
05:50.55jblackIt says that the call ended up in the default context, but it couldn't find '+1210....
05:50.57drmessanoUh no
05:51.08esaym_+1NXXNXXXXXX/114 should be valid?
05:51.22SwKesaym, are you playing w/ level3?
05:51.30esaym?
05:51.32jblackesaym: Only if your callerid is 114.
05:51.49jblackdrmessano: It's not just sip that's te problem though.
05:51.52SwKL3 is the only company i know that forces the + in front of e164 numbers
05:52.06*** join/#asterisk Maxous (n=Maxous@76.97.3.24)
05:52.09jblackIf you open up 1k ports, then other things that just happen to end up in the range are accessable as well.
05:52.27esaymyes see bottom of the pastbin for the sip.conf with the callerid= stuff
05:52.32iamthelostboyif i forward a bunch of ports to the asterisk server, and another application decides it wants to use those ports, the users suffer?  or am i misunderstanding how the firewall will handle it?
05:52.39esaymit was set to user <114>
05:52.56jblackesaym: Just becaue you meant to set it doesn't mean it was actually set.
05:53.05jblackcheck your callerid log and see what was reported
05:53.14UnixDogdid 1.6 go beta
05:53.16jblackpardon, your call record log.
05:53.21drmessanoYes it did
05:53.22jblackUnixDog: It did.
05:53.29UnixDogok
05:53.30esaymso caller id was  114 so i should have used the exten => _+1NXXNXXXXXX/114,1,Set(CALLERID(number)=2108810678)
05:53.39UnixDognow I have to grab and port
05:53.39jblackUnixDog: So make sure you keep at least 1" of lead between you and 1.6.
05:53.53SwKjblack, you know that you can control what RTP ports you are using... and if you are using them for RTP then you shouldnt be using htem for some other application...
05:53.54jblackessaym: Can you show me the call record?
05:53.54UnixDog?
05:54.03UnixDogwhats wrong with it
05:54.09UnixDogany major issues
05:54.14jblackunixdog: It's beta!
05:54.16[TK]D-Fenderjblack, no, beta radioation is blocked by 1' of WATER
05:54.25[TK]D-Fenderjblack, that'd be for GAMMA
05:54.38jblackMaybe that's why I got the brain tumor. :)
05:54.38UnixDoglol
05:54.52UnixDogdid zaptel go 1.6 also
05:54.57UnixDogand libpri ?
05:55.01[TK]D-Fenderjblack, You should ask Dubbya for a course on nucular fizix!
05:55.03russellbnot zaptel
05:55.04jblackunixdog: YOu do know how to /topic, right?
05:55.13russellbyou use zaptel 1.4 with asterisk 1.6
05:55.17russellbeverything else went to 1.6
05:55.33UnixDogok
05:55.36esaymjblack: yea no pb
05:55.49UnixDogwell it will get tested tomarrow
05:55.57UnixDogand patched where needed
05:56.17UnixDogand a port made
05:56.27drmessanojblack: If you see MrDigital, you need to let him know he needs upgrade that school to 1.6 ;)
05:56.27jblackis that "Yeah, no problem", or "Yeah, no pastebin" or.. ??
05:56.29russellbok ... well, i don't expect problems, given that we have a core FreeBSD developer that is also a very active asterisk developer ...
05:56.44jblackdrmessano: I'm sure he already put them on 1.6.4 by now.
05:56.48drmessanoROFL
05:56.53drmessano1.8 Beta
05:56.55UnixDog?
05:57.03jblacklong story unixdog.
05:57.07drmessanoyeah
05:57.11SwK1.2 foreva
05:57.18drmessanoIt starts underneath a Tennis court
05:57.19jblackLet's just say it involves cheeseburgers and tennis courts.
05:57.20drmessanoand goes from there
05:57.25UnixDog1.2 eol rest in pieces
05:57.35SwK1.2 > 1.4
05:57.36jblackcant. breathe
05:57.38drmessano"Would you like BGP with that?"
05:57.47jblackstop dizzy
05:57.53UnixDog1.4 rocks but hoping 1.6 is better
05:58.06SwK1.4 falls over too much for me heh
05:58.12esaymhttp://pastebin.ca/862494
05:58.32UnixDognot had a issue yet
05:58.32esaymyes funny, doesn't seem to have the extension number as the callerid when it dials out?
05:58.33SwKand the changes they forced on the UI for trunk/1.6 I completely hate
05:58.42UnixDogbut then again on freebsd we patch things to work
05:58.44jblackI did notice some problems with 1.4 in the last week or two. They went away once I made sure I had a clean dialplan.
05:58.47drmessano"I'll take a layer 3 roast beef and cheese with extra ARPys sauce"
05:59.10jblackYou're gonna go to jail for murder
05:59.14drmessanolol
05:59.35[TK]D-Fenderjblack, but he's running EXT3!!!!
05:59.53drmessano"hey man, can you reboot that switch.. I gotta flip these burgers.. the buns are getting too warm"
06:00.29jblackMrDigital?? No way! I bet he's running MagicDistFS, that can get 12GB/sec on a P-II/300, over an accoustic modem
06:00.53[TK]D-Fenderjblack, you missed the "murder" joke in there....
06:01.12drmessanoHes got a 1.21 Gigawatt processor
06:01.14jblackhere, google boy
06:01.36jblackOh, You're thinking reiserfs
06:01.41drmessanolol
06:01.47Nuggetthat's the one that destroys your data, right?
06:01.52jblackAnd your ex-wives.
06:01.59drmessanoNo, it buries it
06:02.08jblackThere's no proof that she's buried.
06:02.25[TK]D-Fenderjblack, reiser5 isn't Journaled ;)
06:02.30jblackHe could have stuck her in a lake, or inside a hollow tree... perhaps boiled her away with acid.
06:02.46jblack[TK]D-Fender: Correct. reiser is not journaled, but incarcerated.
06:02.52drmessanoExcuse the hell out of me.. it takes your data and runs away to russia with it, blaming you for the data losss
06:02.57jblackHow's the trial going, anyways?
06:03.21[TK]D-Fenderjblack, like molassas going uphill in January...
06:04.47jblackHere's proof he didn't bury her:
06:04.52drmessano"i told you i'm stable, no need to RAID my place"
06:04.53jblack<PROTECTED>
06:05.08jblackUnless he bought three shovels and two picks....
06:05.41jblackdrmessano: lol
06:05.48drmessanoSo he had her killed.. Err umm.. excuse me.. he had her murder forked
06:06.01esaymhmm, yea I am at a loss with the broken exten.  It's late
06:06.18jblackesaym: I'm still waiting for the cdr record.
06:06.28esaym0_o
06:06.35jblackHey, to you and I, it's murder. To him, it's a simple unlinking
06:07.08jblackThats what all the confusion is about. A simple case of non-converging vocabularies
06:07.16esaymjblack: are you archiving my logs to cdr's to sell on ebay?
06:07.36jblackYou're right. I'm evil. You shouldn't tell me.
06:07.41esaymrofl
06:07.50[TK]D-Fenderjblack, she had a framentation problem so he put her in the recycle bin ;)
06:08.05jblacklol
06:08.10jblackI need a smoke.
06:08.27jblackAnother good piece of news. That ugly hand on namesys.com is gone
06:10.09drmessanolol
06:11.25drmessanoHmm
06:11.33drmessanoMaybe he wanted to /swap and she didnt?
06:13.04jblackaahhh. The simple pleasure of inhaing someone after it's been set afire.
06:13.08esaymwell I got to get in the shower, but if anyone figures it out I will be back in about 20 minutes
06:13.45drmessanoIs that a threat?
06:13.49jblackesaym: Odds are your calling client is either giving different cid than you think, probably by falling into the wrong sip context.
06:14.20jblackmy brilliant plan involved getting him to look at the call record and having an "AHA" moment...
06:14.43esaymyea I am wondering, the call log shows nothing at all for originating id when a big number is dialed
06:14.51jblackbut he more brilliantly still foiled that with a NoOp()
06:15.02jblackesaym; That's your problem. Bad callerid
06:15.47[TK]D-Fenderomg, I see it...
06:16.04[TK]D-Fenderthis is funny...
06:16.05jblackYou see the hand?
06:16.13[TK]D-Fenderjblack, no esaym's problem
06:16.15esaymwell that is the call log, the verbose shows it:
06:16.31jblackdid he pastebin and I missed it?
06:16.32[TK]D-Fenderesaym, -- Executing Set("SIP/114-007bb640", "CALLERID(number)=2108810678") in new stack
06:16.51jblackYup. that would do it
06:16.57esaymbut the regex was before that right?
06:17.03[TK]D-Fenderesaym, See that?  It executed a line of dialplan shoing it accepted your call (which we would have ACTUALLY seen if you showed us the FULL call)
06:17.12Shaun2222sweet, i seg faulted asterisk.... :/
06:17.36[TK]D-Fenderesaym, Well guess what.. you can't match the next priority for your exten... because you just changed the damned callerid!
06:17.46[TK]D-Fenderesaym, You ripped the carpet out from under it!
06:18.26esaymrofl!
06:18.32esaymhmm
06:19.04esaymso try:
06:19.04esaymexten => _NXXNXXXXXX/113,1,Set(CALLERID(number)=2108810677)
06:19.04esaymexten => _NXXNXXXXXX/2108810677,2,Dial(SIP/${EXTEN}@cnm)
06:19.28jblackThat is so evil.
06:19.37[TK]D-Fenderesaym, Time to start using macros and other contexts
06:19.56esaymlol, yea I know, I am learning
06:20.02[TK]D-Fenderesaym, and no, your dialplan is already too bloated and broken
06:20.22esaym:-/ was like 4 days of work
06:21.02jblackAsk yourself for a refund.
06:21.54[TK]D-Fenderesaym, And CID matching is crap.  you should have jsut put your extens into separate contexts so that only the appropriate devices could call them anyways
06:22.18[TK]D-Fenderjblack, if you don't like our free help we'll give you DOUBLE your money back!
06:23.37drmessanoI WANT MY WEEKEND BACK
06:23.42[TK]D-Fenderjblack, I honestly didn't spend even 5....
06:23.43esaym[TK]D-Fender: Now that is brilliant!
06:23.48esaymdidn't even think of it
06:24.03drmessanoPay your bill at the door
06:24.15jblack[TK]D-Fender: Not from you. ;)
06:24.16drmessano[TK]D-Fender will be your cashier
06:24.29[TK]D-Fenderdrmessano, have you learned your lesson?  Oh... and I haven't seen him all week!
06:25.25esaymhey well thanks guys.  I will work on getting some new contexts  out instead of using matching
06:25.37jblackGood idea.
06:25.59esaymit will only be slightly bloated then
06:26.09esaymgood night all
06:26.33drmessanoWell, my only mistake was underestimating how much a user can F**K up a system.. I thought in 10 years of IT I had seen it all
06:26.36jblackwtf. Jewish rap?
06:27.52[TK]D-Fenderjblack, welcome to what... 1983?  Beastie Boys <-
06:28.04jblackHeh, beastie boys this ain't.
06:28.15jblackThis is a rap about bagels and lox.
06:28.35[TK]D-Fenderjblack, I doubt very little... especially the bizarre.
06:28.49[TK]D-Fenderbut its time for me to check out.  later all.
06:28.52[TK]D-Fenderzzzzzzzzzzzz
06:28.54jblackSleep well
06:30.17*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
06:30.21jblackOh, good ole fashioned devil worship.
06:30.54*** join/#asterisk brendanpuck (n=brendan@126.54.233.220.exetel.com.au)
06:32.32jblackheh. Here's an artist called "Professor armchair". Is dont-even-try en vogue?
06:34.06Maxous?
06:34.14jblackI'm going through magnatune
06:34.30MaxousHope everyone has a wonderful weekend.
06:34.34MaxousNight Everyone.
06:34.41jblackWhich seems to be the walking wounded of fatally flawed bands.
06:34.44jblackYou too.
06:42.12*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
06:46.02*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
07:18.11jblackOhh, The first business executive jumped out of a window.
07:18.26jblackMARLTON, N.J. (AP) — An executive of a collapsed subprime mortgage lender jumped to his death from a bridge Friday, shortly after his wife’s body was found inside their New Jersey home, authorities said.
07:22.39drmessano2008 yay
07:26.38jblackhmm?
07:26.44jblackI thought you went to bed?
07:27.00drmessanoNo, been playing with this jabber stuff
07:27.13jblackhow's that coming along?
07:27.35drmessanoPSi wont connect
07:27.45drmessanoFigured I would try a client
07:28.53jblackhowabout gaim.
07:29.02drmessanoDunno
07:29.11drmessanoPSi should work
07:29.12jblackThat does both gchat and aim... and about a zillion other protocols.
07:29.15drmessanoIf it will work at all
07:29.45jblack(my thinking is anything that supports anything probably understands specs well)
07:30.20drmessanoWell, it only has to know 1 to work.. lol
07:30.31drmessanoand if PSi wont work I doubt asterisk will
07:31.20drmessanoTheres got to be something going on with it
07:31.25drmessanoMaybe it got slashdotted
07:31.45jblackYeah. I'm surprised you're jumping on it so fast.
07:32.03J4k3so you guys hear time warner is trying out METERED service
07:32.06J4k3in beaumont, tx
07:32.19jblackYeah, in a place full of old people that are lucky to use 5 megs a day.
07:32.31J4k3nah, beaumont is too violent to have old people.
07:33.32drmessanoHere is the part I love
07:33.33jblackHey, I found music that doesn't suck
07:33.58drmessanoThey somehow justify it by saying 5 percent of their customers use 90% of their bandwidth
07:33.59drmessanoWell
07:34.14drmessanoThen they're overcharging 95% of their customers
07:34.18drmessanoWAIT, WUT?
07:34.35jblackhmm. That's a good point.
07:34.55drmessano95% of their customers deserve 90% off their bill then
07:35.01J4k3yaeh
07:35.05drmessanoHmmm.. wait
07:35.06J4k3and thats not whats going to happen
07:35.10J4k3TW will make more off this scenario
07:35.13drmessanoIs that the sound of them STFU'ing?
07:35.47drmessanoWell, Hillary and Obama will save us
07:35.48J4k3its just a way to kick off abusive users
07:35.50J4k3hahaha
07:35.54J4k3right
07:36.03drmessanoRon Paul for secretary of the UN!
07:36.05J4k3hillary sucks... well no she doesn't, thats where monica comes in
07:36.10J4k3rupaul
07:36.25drmessanoMy wife said "Wouldnt it be cool if Hillary ran with a woman"
07:36.34J4k3well
07:36.34drmessanoI said "Shes been running with women for years"
07:36.40J4k3the democratic party has split themselves into a no-win
07:36.42jblackI _really_ hope hillary doesn't win.
07:36.51J4k3a black muslim vs white chick w/ oral issues.
07:36.52drmessanoTrue
07:37.06jblackShe wants to get a law that makes buying health insurance mandatory.
07:37.27drmessanoVoting for a woman that wont put out sets a bad example for the women that refuse to get barefoot and make babies
07:37.32J4k3I've already seen what the manditory drug plan for medicare has done
07:37.33drmessanoWHERE IS MY POT PIE?
07:37.40jblackCan't pay your bills, can't heat your home, can barely feed yourself, and now you have to .
07:37.41J4k3jack shit, cheaper to get scripts at walmart for $4/script
07:38.36drmessano$4 drugs at Wal Mart x the millions that dont need them
07:39.09drmessanoI feel shoopy now..
07:39.56jblackI hope you cut the arms off your t-shirt after you said that.
07:40.17jblackPehaps get yourself a mullet and knock out one of your front teeth.
07:41.16drmessano"Do you feel a peeing sensation when you pee?  Do you get headaches listen to loud music?  Does heat make you sweat?"
07:41.37drmessano"Here take two of these... then one every day forever"
07:42.46drmessanoI started taking Ginseng and Ginko in the morning... and now I dont remember why.
07:42.49drmessanoI guess it's working
07:43.30jblackheh
07:44.45jblackToday I asked my kid if she believes in UFOs. First she said no. Then she elaborated to "I think there's life all over the universe, but I don't think the world is being stalked"
07:44.54jblackThat made me happy.
07:45.03drmessanolol
07:45.13drmessanoSaves on the tinfoil
07:45.33jblackaye. I need all of it I can get my hands on to keep out the NSA.
07:45.47jblackAnd the Feds, and the CIA, and whoever it was that killed kennedy.
07:45.59drmessanoKennedy was a suicide
07:46.07jblackOhhh. Good one!
07:46.29jblack"The magic bullet was carved out of a boomerang"
07:46.51drmessanoPop rocks and coca cola
07:47.34jblackThat explains why he doubled over.
07:48.26drmessanohttp://dopeman.org/xmpp_srv_test/
07:48.33drmessanoThats a cool site
07:48.45drmessanoIf you configure an XMPP box, it checks your SRV records
07:49.11jblackWhat is xmpp/
07:49.27jblackHold on here.
07:49.34drmessanoJabber == XMPP
07:49.38drmessanoDamn
07:49.41drmessanoSite isnt working now
07:49.53jblackYeah. Everything I look up is blank
07:52.13drmessanocrap
07:53.54jblackIM BORED
07:54.06drmessanoIm annoyed
07:54.23jblackI'd be happy with a "How many asterisks can I put on my 386" question right about now.
07:54.46drmessanoASKERISK MAEK FREE CALL?
07:56.01jblacksure. it works with tennis courts.
07:56.25jblackwhoah. check /.
07:57.05jblackForget tinfoil. I'm going to line my house with batteries.
07:57.13drmessanoYep... was just reading that
07:58.15jblackgreat. so now I have to get a generator.
07:59.47drmessanoI trust the CIA
07:59.59drmessanoThey did so well with 9/11
08:00.15jblackYeah, they did great with the WTC both times.
08:00.22drmessanoI bet 9/11 wasnt planes, but incadescent bulbs
08:00.47jblackIt's a plot to get everyone to put CFLs in their home, and mercury poison their children.
08:01.11jblackIt's a long term plan to keep republicans in office.
08:01.49tessierNonono...IT'S THE QUEERS! THEY'RE IN IT WITH THE ALIENS! THEY'RE BUILDING LANDING STRIPS FOR GAY MARTIANS!!!
08:01.59jblacktessier!
08:02.03tessierjblack!
08:02.08tessierHow's it hangin?
08:02.13jblackI was just thinking about you today. Wanted to talk with you about dundi.
08:02.27jblackPretty comfortably. The new briefs help
08:02.31drmessanolol
08:03.13tessiercool
08:03.17tessierUnfortunately I don't know much about dundi
08:03.36jblackI can give you the concept in 10 words or less.
08:03.57jblackI might even be able to do it in a SAT-style analogy
08:04.43jblackdundi is to telephony as magnatune is to music.
08:04.52tessierWhat's magnatune?
08:05.14jblackMusic site where people publish themselves. Anyone can listen, you can donate if you want.
08:05.26tessieroh
08:05.31tessierThat sounds familiar.
08:05.37jblackyah, it should
08:05.37tessierI used to work for an outfit like that.
08:05.45jblackyup
08:06.02tessierI know that much about dundi. But I have never set it up or anything.
08:06.03jblackbasically, peers agree to share routes with other peers to PSTN numbers.
08:06.21tessierSounds like a good place for a kademlia style network with phone numbers/routes as keys
08:06.29jblackI'm part of a small dundi network that is serving chicago, a big chunk of PA, Conneticut(sp).
08:06.45tessierCool
08:06.53tessierI need to get to bed now but I'll catch you later and we can chat more about it
08:06.54jblackWe could always use more members
08:06.58tessierI'm game
08:06.59jblackOk. Sleep well
08:07.01tessiernight
08:08.17jblackhuh. fn-window pulls up the shutdown box.
08:09.29jblackdrmessano: Lets bump up the excitement in here. Go grab your scissors, and we'll run about for a bit
08:09.37jblackFirst one to poke out an eye wins
08:09.54russellb~thwack jblack
08:09.55jbotACTION smacks jblack on the ear with a AS/400
08:10.15jblackYou're still here too?
08:10.23russellbsort of
08:13.18drmessanolol
08:14.45jblackOhh, I know.
08:14.55jblackLet's 3 way dial fidel castro and the white house. =)
08:15.14*** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it)
08:15.20drmessanoHmm
08:15.30drmessanoI wonder if the switchboard @ 1600 is on ENUM
08:16.15jblack202-456-1414
08:16.33drmessanoyeah
08:16.43jblackyeah, that's it, or yeah, it is
08:16.53drmessanoThats the number
08:16.57drmessanoNot sure about ENUM
08:18.24drmessanoenumquery.com says "no"
08:18.30jblackdurh.
08:19.40drmessanolol
08:19.53drmessanoWest Coast wakeup calls
08:20.30jblacknah, just lookups
08:21.07drmessanolol
08:22.09jblackevery 800 number is in there.
08:23.04jblackI know. I'll check O'reilly media.
08:23.18drmessano800 works well
08:23.19jblackIf they're not enumed, we can lart some people here. =)
08:23.23drmessanolol
08:23.24jblackThey have a 707
08:23.43jblackWoot. Failure.
08:23.55drmessanoWow
08:24.08jblackLoose-ee: You're in trouble!
08:24.16x86efnetgive me fidel castro's phone number
08:24.28jblackLessee. Jim van Meggelen, Leif Madsen & Jared Smith
08:24.43jblackhow would I know fidel castro's number?
08:25.55jblackcall up the cia (I'm sure they're listed w/ the whitehouse switchboard) and ask them for Fidel's number so that you can prank him.
08:26.07jblackWho knows. they just might give it to you on the promise you'll annoy the hell out of him
08:28.14drmessanoGoogle is of no help
08:28.54jblackOk, here's the US Interests Section in Cuba (Vedado, Havana)
08:29.01jblack53-7-833-3551
08:29.51jblackThey should know how to get ahold of the central committee, which is their version of parlament, I believe.
08:30.45drmessanoHmmm
08:30.51drmessanoSo umm
08:31.02drmessanoIf send all calls from cuba to a context
08:31.12drmessanoCan I use in-fidel
08:31.34jblacksure, I dont' see why not.
08:31.54drmessanoIrony..
08:32.08Shaun2222well background/waitexten seams to work with dial+gosub
08:32.12Shaun2222thats good
08:32.27drmessanobrb
08:32.27jblackFidel.. speaking of the CIA and it's quality work...
08:32.41jblackShaun2222: Pardon?
08:33.52jblackx86efnet: I hope you undersand we're just kidding around. Communicating to/from cuba is likely to get you onto all sorts of lists you dont' want to be on.
08:34.49jblackOh.. NSA, since you're watching, can I have a helicopter-black coffee cup? Please?
08:36.22Shaun2222whats with gosub, does a return always need to be at the end of a sub?
08:36.23x86efnetlol.
08:36.39jblackShaun2222: You should be able to sprinkle in returns as appropriate.
08:37.24Shaun2222GOSUB_RESULT doesnt look to be read properly either...
08:37.39jblackWait. were you asking, do I need at least one, or can I have more than one
08:38.09Shaun2222wondering if a return has to always be the last thing executed in a sub.
08:38.45jblackwell, the first return your logic hits will certainly be the last thing executed for the gosub.
08:39.27jblackIf you don't explicitely return, I imagine you'd fall out of the dialplan as normal.
08:39.40jblackOr do a terminating action like a Successful Dial
08:54.04Shaun2222cource 1.6 offers features i could now :/
08:56.41*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
08:56.41*** mode/#asterisk [+o russellb] by ChanServ
09:01.33*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
09:06.58*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
09:16.20*** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk)
09:19.48*** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se)
09:21.25jblackDoes anyone know how to get the PSTN for a PSAP?
09:31.12jblackI know my PSAP is at the local police station. Somehow, though, I suspect they won't understand me if I ask them for the PSAP PSTN
09:34.12BBHossyou mean the DID?
09:35.13BBHossYou might try calling 911, then hang up, the caller id might be a direct dial number (this is somewhat illegal though)
09:36.38jblackMy provider is based in colorado. I'm pretty sure that they don't know which PSAP to hook me up with.
09:36.48*** join/#asterisk razor (i=razor@rapwap.razor.dk)
09:38.04razorIt seems that the Record application doesn't make asterisk send any RTP-packets while recording, which is a problem because my upstream provider disconnects when no packets is received after 30 seconds.
09:38.27jblackthat sucks
09:38.38razorCan you somehow send some rtp keepalive packets in the background?
09:40.50jblackYou could try forging a 0 length packets.
09:41.44*** join/#asterisk frek818 (n=herman@adsl-69-234-203-190.dsl.irvnca.pacbell.net)
09:43.07razorseems there is a rtpkeepalive setting in sip.conf - would that work?
09:48.58russellbthere is an asterisk.conf option that enables transmitting silence while recording
09:49.35russellbtransmit_silence_during_record = yes
09:49.40russellbin the [options] section
09:50.04razorthanks, i will look into that
10:00.44razorit worked, great :)
10:03.34*** join/#asterisk sergey (n=sergey@91.189.233.71)
10:05.10*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
10:10.47*** join/#asterisk tasterisk (n=keyser30@rrcs-24-73-85-186.se.biz.rr.com)
10:12.50yangis anything wrong with this config, I can dial out but i cannot be reached inbound onto the number (059213013)- http://openpaste.org/en/4729/
10:13.11jblackprobably a context issue
10:13.28jblackset verbose 99, and try dialing again
10:13.52jblackIt'll probably say something like "no such extension 059213013 in context default"
10:15.44jblackif you still don't see it, try set debug 99, and try again.
10:15.56jblackand if you _still_ don't see it, then type sip debug on, and try a third time.
10:15.58yangyou mean i need to put "verbose=99" into [detel] ?
10:16.15jblackI mean run "asterisk -r", which will put you into a console.
10:16.27jblackthat's what we use to watch what's happening.
10:16.28yangyeah i am in asterisk -vvvr allready
10:16.38jblackOk, there, type "set verbose 99"
10:16.42yangok
10:17.13jblackdial in and watch carefully for what I said.
10:17.34jblackif verbose 99 doesnt show it, debug 99 should. If that doesn't sip debug definitely will.
10:18.59yangwell its really strange, maybe DID isn't linked to my account from the VOIP provider site ?
10:19.21yangverbose 99 hasn't given me anything
10:20.14yangOr maybe there are some special strings required from the voip provider to work?
10:21.20*** join/#asterisk ghtdak (n=tarbox@c-24-19-22-67.hsd1.mn.comcast.net)
10:21.50yangyou know when i dial to Detel inbound number from my mobile...i get signal "no number exist - weird beeps"
10:37.50*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
10:51.07yangjblack: any more suggestions?
10:59.06*** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled)
10:59.58slimahuh, 1.6!
11:00.01slima;-)
11:00.12troubledmoin
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11:03.43troubledanyone know if iaxy's support g.729a?
11:09.51jblackhmm
11:10.20jblackyang: what happened when you ran sip debug?
11:10.37*** join/#asterisk javar (n=javar@static-adsl201-232-227-235.epm.net.co)
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11:19.29jblackYay hardy heron. Two crashes in 24 hours
11:21.00mvanbaakit's because of the name
11:21.24jblackMust be.
11:21.39mvanbaakI mean, who names their release hardy hardon
11:21.40jblackperhaps MrDigital is in charge there
11:21.57yangjblack: there were some messages scrolling
11:22.08jblackOk, so they're ringing you.
11:22.18yangjblack: but i don't know if they were related to the rings
11:22.32jblackOk. Hit enter a bunch of times, then call again.
11:22.34yangshall i paste the debug somehwere?
11:22.49jblackIf it's ringing in, you'll get a flurry of messages. I want those.
11:23.09jblackalso, do a dialplan reload, and look carefully at the copious output for warnings.
11:24.20jblackIf you still can't find it, paste (in this order) , the following three things: 0. a reload  1. your sip.conf (with passwords renamed)  2. your dialplan (with passwords renamed), 3. the sip debug output after the reload
11:24.52jblackSome day real soon I'm going to put a page up on the wiki that clearly explains needed info.
11:25.11jblack~debughelp
11:25.30troubledso, anyone know about the iaxy and firmware updates?
11:29.01yangjblack: this happens when i am trying to ring - http://openpaste.org/en/4737/
11:29.54yang~debughelp
11:30.07jblackuhh. hrmm.
11:30.09jblackjbot isn't jbot
11:30.13jblack~pb
11:30.14jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
11:31.40yangjblack: is the incoming call visible on that "sip debug" ?
11:31.55jblackIt should be, yes.
11:34.03tzafrir_hometroubled, ask a more specific question. That said, I don't know much about the iaxy
11:34.03yangthere are also a bunch of other voip provider messages, ignore voip.ms and voipbuster
11:34.03jblackjust give me the whole mess.
11:34.07yanghello tzafrir_home
11:34.42jblackjbot is acting rather odd. It's kind of laggy, and it answered me when I asked it how it was doing.
11:35.03troubledtzafrir_home: thanks. i didnt realize that my iaxys supported firmware updates. i figure they were just refering to iaxprov "updates". im rather interested if they added anything new to the firmware and where i might find out more about it
11:35.29troubledalthough ill be here for tdm questions as well, so dont go anywhere ;)
11:35.46tzafrir_hometroubled, look for iaxprov.conf and its sample
11:35.49jblackyang: Making progress?
11:35.50tzafrir_homemaybe this can help
11:36.32troubledtzafrir_home: they have already been provisioned, connected and work fine, dont recall anything firmware specific though. let me recheck though
11:36.33jblackI setup a ~debughelp1 and ~debughelp2 . Tuning/corrections to it are welcome.
11:37.57troubledtzafrir_home: nah, nothing mentioning "firmware" in svn HEAD for iaxyprov
11:38.24tzafrir_hometroubled, I'm afraid that's just as much as I know. There's a "firmware" directory under /var/lib/asterisk and somewhere under there the iaxy firmware should reside
11:39.11tzafrir_homeI know this only because we had to strip the iaxy firmware from the Debian debs (they dislike firmwares...) and this created some problems at first...
11:41.38troubledactually using debian etch atm. just sitting down to a good 604pg read and noticed mention of that dir you speak of
11:41.38troubledalthough its 1.4 and im running 1.2, that could change down the road
11:41.50jblackOn ubuntu heron, the firmware is kept in 0:mercury:/usr/share/asterisk/firmware/iax#
11:42.06troubledright now i just want to get it all tested. still cant seem to get my fxs (red) module working yet though. but i still got a lot of catching up to do before i get to what might be causing that problem
11:42.22troubledjblack: question is though, where do you get the firmware?
11:42.32jblackwhat device?
11:42.40troubledfor the iaxy
11:43.17troubledi also need to find a way to track down my g.729a license if anyone knows a url
11:46.10frek818Anyone here instrested in a job supporting and Asterisk based PBX system?
11:46.22frek818s/and/an/
11:46.24yangjblack: so, have you diagnosed that openpaste output, is there something related to "detel" account ?
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11:46.50troubledheh, jbot is a riker infobot isnt it
11:46.52yangjblack: or i can put everything in the order that you told me
11:47.14jblackThere is an rpm package with the firmware. You could rip it apart
11:47.34tzafrir_home~riker
11:47.52yangjblack: "dialplan reload" doesnt seem to work for me...
11:48.17jblackOh?
11:48.28yangsip reload you ment
11:48.42jblackWhat version of * are you running?
11:48.45yang1.2.24
11:49.04jblackYou're aware the latest stable release is 1.4.17 ?
11:49.16yangno, i am using debian waiting for the 1.4 to get in
11:49.29jblacktry extensions reload
11:49.53jblackand I did mean sip debug on.
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11:52.07jblacktroubled: You should be able to get it with a svn pull
11:52.27troubledjblack: the firmware?
11:52.42jblackyup. svn pull the sources, it should be within
11:52.52*** part/#asterisk gammos (n=gamma@s15244973.onlinehome-server.info)
11:52.54yangjblack: reload = http://openpaste.org/en/4739/ , making call inbound = http://openpaste.org/en/4737/
11:53.10troubledi grabbed the entire iaxyprov tree, anychance its in there? or are the firmware in a seperate repo?
11:53.39yangmy config = http://openpaste.org/en/4729/
11:54.06jblackIt may be. do a "find . -name 'iaxy.bin'"
11:54.27troubledsec
11:54.46troublednope, not in iaxyprov svn then :/
11:55.37jblackI meant the asterisk sources anyays
11:55.53jblackyang: see line 40 in your first paste?
11:56.05troubledah
11:56.18mvanbaakthe default asterisk tarballs and svn have the iaxy.bin in /var/lib/asterisk/firmware/iax/
11:56.20jblackI haven't looked at the second paste, but the diaplan as posted won't work.
11:56.39jblackYou take the call, then route it to a nonexistant context, which will push an error right back up the line
11:57.23jblackeither that, or the extension they're diling to you doesn't start with 7
11:57.29yangjblack: Contact: <sip:600@212.18.59.220:5060>
11:57.37troubledjblack: ah there it is in asterisk contrib/firmware/iax. ta
11:57.53jblackOk, thre is no line 600 extension in buster.
11:58.03yangjblack: my asterisk server is on 212.103.133.6 , while my phone is inside NAT-ed 212.18.59.220 (home)
11:58.13HowdyDoodyHow do I get "exten=>T,1,NoOp("starting tea section") etc to work after a SET(TIMEOUT(absolute)=19) command ?
11:58.43mvanbaakHowdyDoody: by replacing the T with a t
11:58.45jblackyang: You said they're dialing sip:600@212..., with your detel context, right?
11:58.49mvanbaakexten => t,1....
11:59.13yangjblack: the number that they are dialing is 05921....
11:59.20HowdyDoodyThe dumped the T and reverted to just a "t" now ? (I am version 1.2)
11:59.31jblackOk.
11:59.33yangthat is a PSTN number
11:59.46mvanbaakhhmm, I never used T
11:59.48mvanbaakalways t
11:59.50jblackSo, look at your [detel] context. You can see that calls from this will be set to buster.
11:59.50yangand it should forward the call from asterisk to 600@212.18.59.220
11:59.54jblackNow, let's go to buster.
12:00.08HowdyDoodyOK, I 'll try that and be back later.
12:00.20jblackin buster, you have a wildcard extension (one seven any number of digits after that).
12:00.32jblackWhere is the 05921.... extension that they're trying to call?
12:00.41troubledjblack: quick question though, svn.digium.com is the authorative svn right? just making sure i shouldnt be going to asterisk.org or something in case the digium is some sort of vendor drop or something
12:00.59jblacktroubled: it's about as authoritive as you can get.
12:01.10troubledmost up to date i guess is what im after
12:01.19jblackNo it's not.
12:01.31troubledi would hate to find out that asterisk.org has a svn repo thats months ahead of digiums copy ;)
12:02.04yangjblack: the username 777777 was changed in sip.conf, my username is 631735
12:02.14jblackping: unknown host svn.asterisk.org
12:02.37jblackyang: Don't lie to me.
12:02.57troubledjbot: dns svn.asterisk.org
12:03.07troubledbah
12:03.12jblacklet me look at the other paste
12:03.42yangjblack: ah shit, i am looing at [detel-inbound] its forwarding to extension 400 instead of 600, are you refering to that?
12:03.50jblackSee line 200?
12:03.57jblackpardon, I meant 20.
12:04.33yangline 20 on openpaste 4739 ?Åū
12:04.35jblackYou're on the right track... except what you pasted says they don't go into [detel-inbound] at all.
12:04.41jblackThey're going to [detel].
12:04.54jblackLook at line 13 on http://openpaste.org/en/4729/
12:05.19jblackThat says "Any sip connection though this is going to get dumped into [buster] in the dialplan
12:05.26jblacks/[detel]/buster
12:05.45jblackany sip calls that come in through [detel], if it's not in buster, it doesn't exist.
12:06.18yangso [detel] should actually use context [detel-inbound] ?
12:06.32jblackIf that's what you want, sure.
12:06.43jblackIt would certainly route 059213013
12:06.51yangok i will fix that
12:07.01jblackAlso, you need to delete lines 46 and 46, as they're redundant.
12:07.16jblack(they're already done in 44 and 45, and extensions reload will tell you that)
12:08.13jblackalso, remove the r, ....
12:08.18jblackoh screw this, I'll rewrite it
12:08.57xachenjbot: dns svn.digium.com
12:10.09jblackhttp://openpaste.org/en/4744/
12:10.56jblackyang:  ^----
12:11.00HowdyDoodyThe t,1, did not work either.  See http://openpaste.org/4745/ for more details of what I did
12:11.02yangjblack: so i can start commenting everything from 24-35 (detel-noter)
12:11.22jblackwell, maybe.
12:11.51jblackthat [detel-noter] context is a mess, because of insecure=port,invite
12:11.58yangok
12:12.02jblackI would prefer you disable it, and make sure they're logging in on [detel]
12:12.18jblacknot logging in, but dialing into
12:13.12jblackthere are providers out there that prefer unvalidated connections to validated ones.
12:14.08yangjblack: ok i changed all that....now the signal when calling from my phone has changed....from unexisting number to busy tone
12:14.21HowdyDoodyI see I was using the old syntax, I'll switch to new one and try again.  SET(absolute)
12:14.24jblackThat's great.
12:14.43jblackNow, you need to have an actual extension 600 logged in.
12:15.02jblackWhy don't you replace the DIAL(SIP/600) with a Background(tt-weasels)
12:15.18yanghmmm
12:15.32jblackthat'll make sure the datel connection is working.
12:15.37yangi ll try
12:16.15jblackThen, you can make another sip account that your softphone/hardware sip phone can log into, which you can then Dial() to.
12:16.43jblackCalls come in, and * turns around and Dials the other party and bridges them.
12:16.54jblackRight now, you have only 1 party.
12:17.02HowdyDoodyNewer syntax didn't work any different, phoo. I get call cut off in 19 sec then hear fast-busy tone. Same as before.
12:17.34yangjblack: well, i have just one VOIP handset so i link everything to 600 as this is phone's extension
12:18.06yangjblack: have a good smoke and we can continue later, maybe better in a query
12:21.51jblackFine, but your handset needs to have a sip acount on your asterisk server too.
12:21.58*** join/#asterisk RoyK (n=roy@91.149.21.238)
12:22.07jblackOnce asterisk gets the call, it needs to call your soft phone so that it can bridge them.
12:22.41jblackhere. I'll set you something up
12:25.26yangI need 2 accounts, i understand (bridge mode)
12:28.06jblackyang: done
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12:29.40jblackyang: http://openpaste.org/en/4747/
12:31.08yangjblack: thanks a lot , i will implement it
12:31.16jblackas you get more voip-providers, give them a new sip context, and add their number to inbound.
12:31.32jblackas you get more sip phones, just give them a new sip context.
12:36.32jblackIt may seem odd to put your phones in a [phones] context, just to turn around and include it in the [public] context. It's a good idea, though. As your network grows and you add more stuff, you'll be able to add all of your local phones in.
12:36.50jblackLike say, you make an [internal] context for your inhome network.
12:37.19jblackIf you include => phones to both [public] and [internal], then every phone will always have the same extension.
12:38.53jblackEventually, you'll also want to change line 41 as things get more compicated.. first from _X. to _XXX, then later to individual extensions (i.e. exten => 600 .... exten => 601 ...)
12:48.16*** join/#asterisk yassine (n=yassine@unaffiliated/yassine)
12:55.35yangjblack: my full config files here - http://openpaste.org/en/4749/
12:57.00jblacklooking
12:57.21yangand i am still getting a "busy" tone when calling detel number
12:57.43jblackwhere did all this come from?
12:57.54yangDo you think it could a problem with voip provider filtering their DID's?
12:59.11jblackDid you log your sip phone into asterisk?
12:59.18jblack(you can check with sip show peers)
12:59.52yangits being logged fine, I can also reach myself by calling +1 8773533967
13:00.16jblackso you have an ip for both 600 and detel?
13:00.51yang600/600                    212.18.59.220    D   N      5060     OK (186 ms)
13:00.53yangdetel/631735               80.246.224.110              5060     OK (2 ms)
13:01.26jblackIt could be a lot of reasons. You've got a large number of flags.
13:01.52*** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245)
13:02.04yangWell my assumption is that its being all correct, and that they are probably filtering DID's or blocking it for some reason
13:02.08jblackMy advice is to use the sip.conf and extensions.conf I gave you, and build up from there.
13:02.12jblackI'm going to bed.
13:02.29yangjblack: ok, thanks for your time
13:02.50jblackinsecure=  is bad,bad,bad. Don't use it at all if you can avoid it.
13:03.00jblackUse port if you must.
13:03.52jblackIf you use =yes, or =port,invite, then my limited undersatnding is that _any_ sip device can use it (and often will)
13:08.53*** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr)
13:19.00yassinei have a x100p based card and it was working with asrterisk 1.2 for inbound and outbound calls after an upgrade to asterisk-1.4 two months ago, only outbound is working and asterisk does not seem to notice incoming calls (from Zap) at all. any idea what could be the issue behind this?
13:21.26tzafrir_homeyassine, do you use users.conf in any way? If not: can you pastebin your zapata.conf ?
13:22.44yassinetzafrir_home:  i tried monitoring calls with ztmonitor 1 -vv and dialed my pstn nr and the call is not noticed
13:22.54yassinelet me past my zapata one sec
13:23.39tzafrir_homehmm.... do you see the channel at all in: cat /proc/zaptel/*
13:23.43tzafrir_home?
13:24.58*** join/#asterisk amaache (n=maache76@41.221.16.35)
13:25.16frek818Excellent job opportunity for Asterisk user http://hermangriffin.com/jobs/
13:25.16*** join/#asterisk mrpurple (n=Administ@gre92-7-82-243-130-192.fbx.proxad.net)
13:25.37yassinetzafrir_home: http://rafb.net/p/OUDpJk38.html
13:25.43mrpurplehello i need infos on what i can do with asterisk
13:25.59yassinetzafrir_home: yes i'am able to see the channel using cat /proc/zaptel/1
13:27.23mrpurpleactually i have free as isp (in France) and i have a contract where i don't pay calls in 70 country of the world can i share this ?
13:38.17kodomomrpurple: sure
13:38.34yassinetzafrir_home: do you have an idea for me?
13:43.34mrpurplekodomo: i have a Nas attached to the freebox so i think i can install Asterisk there and share the line ?
13:44.03mrpurplekodomo: do you know the freebox ? mean are you in france ?
13:45.45troubledmy apologies to the svn admin for all the bandwidth use :)
13:48.33tzafrir_homeyassine, is that an X100P? it does not forward polarity events, AFAIK
13:49.07tzafrir_homeSo 'asnweronpolarityswitch = yes' may actually be harmful
13:49.12mrpurplekodomo: free is giving the possibility to use sip protocoll and so i used when i connect with my portable when i move in France. the problem is when i go out of france i cannot use. So i think the best is an Asterisk that give to me the possibility to connect wih the free line. andso i can share with friend ... the problem is i don't know how do it ... :-(
13:49.19tzafrir_homeThough this is mostly for outgoing calls, I guess
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13:49.48*** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk)
13:50.46yassinetzafrir_home: outbound is find well its drops after 6 min if im calling from sip --> zap -> pstn but that is not my primary problem now :s
13:51.41tzafrir_homeRemove the "answeronpolarityswitch and "hanguponpolarityswitch", as your card will not forward those events anyway
13:53.02tzafrir_homeand set busycount to a higher value if you suspect false hangups
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14:35.45ussrbackhi all
14:36.10ussrbackin cli mode i get permanently messages
14:36.10ussrbackRemote UNIX connection
14:36.11ussrback<PROTECTED>
14:36.16ussrbackwhat does it mean?
14:37.33frek818Excellent job opportunity for Asterisk user http://hermangriffin.com/jobs/
14:39.43ussrback?
14:40.54razorussrback, a script/program/user is connecting to the asterisk console
14:42.16ussrbackbut i had not this messages a few days ago
14:42.24ussrbacki did not changed anything in system
14:42.31riddleboxussrback, http://www.google.com/search?q=+Remote+UNIX+connect+++++--+Remote+UNIX+connection+disconnected+Remote+UNIX+connection%0A%3Cussrback%3E+++++--+Remote+UNIX+connection+disconnected&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a
14:42.43ussrbackhow can i find what s trying to connect
14:42.50riddleboxsorry I didnt know know that link was that long
14:48.06kodomomrpurple: did I get you right that you use regular SIP with your provider account? Then all you have to do is make asterisk register on your behalf, create a user account on your own box, to which you register and associate that account with a context, in which you use your provider for everything you dial...
14:49.08kodomomrpurple: if you don't know it yet, there's a whole lot of documentation, howtos, tutorials etc. on how to accomplish that on voip-info.org
14:51.01*** join/#asterisk nybble (n=nybble@about/apple/performa/nybble)
14:51.56ussrbackhow can i fix this error ? -->  pbx.c:1553 ast_func_write: Function Set not registered
15:01.13tzafrir_homeussrback, functions are upper case
15:01.22tzafrir_homeuse SET isntead of Set?
15:01.39tzafrir_homeOr maybe you wanted to use the application Set?
15:02.08ussrbackwhere should i change ?
15:05.10*** join/#asterisk anthm (n=anthm@68-31-59-195.area4.spcsdns.net)
15:05.10*** mode/#asterisk [+o anthm] by ChanServ
15:07.35troubledgah, you guys could have told me the full asterisk svn weighs in around 10 gig or so heh
15:08.14troubledjbot: lart the team svn dir
15:08.14jbotacting on orders from an unspecified client drags the team svn dir into court suing for $200 million
15:08.33troubledheh, should cover my bandwidth costs almost ;)
15:09.14*** join/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk)
15:11.51mrpurplekodomo: yes i already use sip account from my provider and others. the fact is that my provider is giving my free calls if i call from french ip. when i go in other country i cannot. so  now i'm tryng to install an asterisk in my nas and then i'll see the web site you give to me  .. thank you
15:14.33*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
15:15.35troubledmrpurple: oh? your nas is just a standard linux OS powering it?
15:16.03troubledlike a 4U nas? or just some homebrew boxen with tons'o'drives?
15:23.53Lawbringerhello people which pastebin should i use?
15:24.05Lawbringeroh this is asterisk :S
15:24.51tzafrir_homeLawbringer, anyone you like
15:24.58*** join/#asterisk SteveTotaro (n=SteveTot@pool-70-16-26-249.balt.east.verizon.net)
15:25.16tzafrir_homepastebin.ca has support for asterisk .conf syntax hilighting.
15:31.15DarKnesS_WolFnew asterisk out :D
15:31.16DarKnesS_WolFwow !
15:31.21DarKnesS_WolFtzafrir_home: hello dude how are u ?
15:31.52tzafrir_homeDarKnesS_WolF, all looks well so far...
15:32.21mrpurpletroubled: my nas is a synology ds106j
15:32.43troubledmrpurple: never heard of it, but ill google it in a sec
15:32.51mrpurpletroubled: you can see specification jere http://www.synology.com/enu/products/DS106j/spec.php
15:33.41mrpurpleis a great device .. is doing as server ftp and also actins as a torrent and p2p ..
15:34.14mrpurplei bought some yrs ago ..
15:34.32troubledahh, just a single hd then
15:34.49troubledi was thinking you had some massive 4U multi TB setup or something :)
15:35.26mrpurplenot yet :-D
15:35.37troubledi dont like enclosures without fans though. i had a 160g drive near melt the paint off the drive. and my one had a fan!
15:35.45mrpurplethe problem is my life .. shared btw France and italy ..
15:36.05troubledshared france and italy?
15:36.13*** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it)
15:36.51troubledbtw, wasnt there some UPNP sploit in the wild again from a few days ago?
15:36.57mrpurpleyeah . so i need to have server on all the time
15:38.01mrpurpletroubled:  i hate fan noise .. so i choose one without ..
15:38.39mrpurplemine is 500 g and is fine .. never had problem
15:38.47troubledone i had was so small you could barely hear it. but after taking the hd out of that, i soon realised that enclosures may not be the smartest idea for important single copy storage :)
15:39.05*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
15:40.08mrpurplenow i'm tryng to configure asterisk . .. following this guide ... http://www.voip-info.org/wiki-Asterisk+Slimming
15:40.30troubledheh, you should check out this beast im into atm :/
15:42.29mrpurpletrobled is missing something ?
15:42.29troubledits some 600 pg pdf
15:42.55troubledsorry, cant copy/paste the url with my setup atm. but its from www.asterisk.org, some future print orielly book for 1.4 or somethin
15:43.04mrpurpleehehhhe thank you
15:43.28troubledwhy are you thanking me?
15:43.40troubledno what I do?! ;)
15:43.50mrpurpleinfact the problem now is that i don't know what module i have to activate ..
15:44.00troubledzaptel?
15:44.15mrpurplethe link .. i'm looking for the pdf
15:44.28troubledsec
15:44.44troubledwww.asteriskdocs.org/
15:44.54troubledclick the book :)
15:45.26mrpurplethat is great
15:45.32troubledjbot: listkeys asteriskdocs
15:45.47troubled:/
15:46.25troubledjbot: listvalues asteriskdocs
15:47.00yangwhich linux client is recommended for asterisk use - Ekiga ?
15:47.23mrpurplealla the pakage i installed on my nas are from nslu ... so i start form here
15:47.42mrpurplehttp://www.nslu2-linux.org/wiki/Optware/Asterisk
15:47.46troubledyang: client?
15:48.01af_softphone sip
15:48.05troubledah
15:49.22mrpurpleyang in ubuntu i'm using ekiga and is fine ..
15:50.36*** join/#asterisk admin0 (n=admin@116.90.228.34)
15:53.51yangI used to work with ekiga, i was wondering if any other client (tested) exists
15:54.45SwKxlite
15:55.33mrpurpletroubled: now asterisk is running ... on nas (but don't kno how use it, yet)
15:56.09troubledmrpurple: lots of vim /etc/asterisk/* :)
15:56.27*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)
15:56.40troublednot sure how many calls you could get on that 200mhz cpu in that nas though. plus all the other overhead
15:57.43*** join/#asterisk d3wayne (n=deeewayn@76.29.245.9)
15:57.43*** mode/#asterisk [+o d3wayne] by ChanServ
15:57.57mrpurpletroubled: i need only one at time .. just that i can use this line when i'm in italy .. to make free call allover the world ...
15:59.30troubledmrpurple: very handy. i got an iaxy for something like that myself (remote phone calls)
16:01.05mrpurpletroubled: seems great .. i didn0t knew it
16:01.30troubledya, neat lil things
16:02.14troubledalthough not being able to use dns for connections is a big limitation. kinda forces you to manually update in the field or use vpn for everything
16:02.25*** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it)
16:03.26mrpurplethe problem is im ignorant on all this stuffs
16:03.38mrpurpleso i need help and time
16:03.48troubledi aint far ahead of ya then ;)
16:04.49mrpurplethank you  ..... may i ? :D
16:05.11troubledbut if your up for a challange, you could always just take something like this asteriskNOW iso i read about, rip out the cgi/guts and pop it into your nas for a fancy premade setup :)
16:06.15mrpurpleasteriskNOW is going in windows directly ?
16:06.25*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
16:06.46mrpurplei'ld like that
16:06.55troubledfrom what i read, its a self contained linux image. sounds like it has a livecd version which can install to a hd as well
16:07.43troubledbut being linux, you can always do a hack job and rip out the cgi stuff, match the asterisk version, compile+install to your nas, then fire up an apache to give you the same exact setup
16:09.30mrpurpleok seems a little complicate .. from where i start .. ? :-/
16:09.34troubleddepends on your nas really. i assume it runs linux? cause you would need to meet/exceed the lib requirements etc. although i suppose static compiles are an option as well for all the libpri stuff. although i dont think you need that lib specifically since your not using any real hardware, but still
16:09.54troubledhmm, well, if you have to ask, you probably best skip that idea </grin>
16:10.09mrpurpleehehhe
16:10.24troubledalthough you could always try install it in vmware on windows. they did mention a premade/vmware ready image
16:10.45troubledits not like you need to actually do anything cept route calls right?
16:10.48mrpurplei have also a linux ubuntu machine
16:11.01mrpurplebut i'm not a linux expert
16:11.29troubledactually, vmware is bad for timing, not sure how bad the clock drift would hurt asterisk, but I know it kills my installs when it comes to time syncs
16:12.17troubledwhy not just put the asterisk install on ubuntu? then setup to allow remote login in case you need to restart your asterisk or whatever?
16:12.27*** join/#asterisk gerhard7 (n=gerhard@195-241-250-146.dial.ip.tiscali.nl)
16:13.05ManxPowerasterisk requires MUCH better timing than NTP could ever provide.
16:13.30mrpurpleyeah i can ..  to learn stuff's .. but ubuntu is my portable ..
16:13.48ManxPowerIn telecom "timing" is really "synchronization", not having much to do with time.
16:13.58kodomomrpurple: welcome
16:14.05troubledManxPower: nod, just the fact that vmware tends to drift bad soon as it looses foreground priority for me. cant see it running that well unless it were realtime or somethin
16:14.19mrpurplekodomo: thaks
16:20.30*** join/#asterisk philippel (n=p_lindhe@c-98-203-245-82.hsd1.wa.comcast.net)
16:21.52*** part/#asterisk Stefan1979 (n=stan@4204ds2-vby.0.fullrate.dk)
16:25.43*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com)
16:41.38*** join/#asterisk angryuser (n=Miranda@df01t2-212-194-105-163.d4.club-internet.fr)
16:41.51troubledquestion, for a pstn connected fxo, should i have callwaiting/threewaycalling enabled?
16:42.27troubledi assume a flash will apply to the fxs and not actually make it to the fxo channel?
16:42.41angryuseri have installed latest version of * but BLF is not working anymore, before i had 1.4.0 any ideas i am using snom's
16:43.01troubledheh, nice nick :)
16:43.53kyrontroubled, fyi, * under VMWare == bad. The problem has more to do with realtime jitter than drift at this point
16:44.05troubledthanks
16:44.18angryuser;) i do not see any REGISTER for BLF messages, any help?
16:44.24kyronangryuser, BLF (Big Loud Farts)?
16:44.38kyronsorry... I need edukation
16:44.53angryuserkyron: it is too late
16:44.59kyrondoeth
16:45.01*** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net)
16:45.12kyronyeah, I'm unrecouperable
16:45.51kyronthat'll teach me to use trixbox "Silly admin, Trix are for kids!"
16:46.37drmessanolol
16:46.39kyronangryuser, out of curiosity, which phones are you using (model)?
16:47.01kyronMCSE experts love Trix ;)
16:47.01angryuserkyron: 320 360
16:47.03*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
16:47.25drmessanoNot only does it suck, they've got a bunch of users with toasted boxes after an update last week
16:47.59kyronHmmm..toasts, goes well with Trix :P
16:48.06drmessanolol
16:48.11kyrondrmessano, ...uhm...I was told Snom were the phones to get!
16:48.17kyron~phones
16:48.17jboti guess phones is http://bani.anime.net/phones/.  While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows:  Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever. places ...
16:48.28drmessanoLot of people like Polycoms
16:48.29kyronoops... ok, so I'll go polycom then ;)
16:48.45kyronNo...Grandstream!...yeah (masochistic whim)
16:49.08kyronThey are Microsoft call center compliant right?...cuz that _has_ to be important/da future
16:50.15kyronTHUD!
16:50.23joatyou forgot linksys btw
16:50.24drmessanolol
16:50.29joatheh
16:50.33kyroncoulors if you're a brit ;)
16:50.43kyronhehehe
16:50.49drmessanoMicrosoft.. Proving that its easy to misread a standard
16:51.02kyronand impose your own..
16:51.07*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
16:51.24joatthe cowbird of applications and standards
16:51.25kyronkinda like the Mythbuster moto "I reject your reality and substitute it for my own"
16:52.01kyronSo I should be Polycom happy...but I find them...uhm...wide, excessively...
16:52.16drmessanolol
16:53.30kyron`Microsoft Office Communications Server 2007` <-- that should plain and simply be illegal (FCC/CRTC wise)
16:54.21kyron"Sorry, your call cannot F"/34%%%1239"!/""***!"/$""" <-- whilst calling 911
16:54.31drmessanoMicrosoft creating an IP PBX is scary... about as much as their IRC server, NNTP server, and first attempt at IM
16:54.37kyronOops...the "server" seems to have been hit by Melissa
16:55.11joatit'll bring new meaning to an application "calling home" too
16:55.27kyronwell, they'll do like they have always, buy out another project and claim it as their innovation... or just plain and simply rip it off one
16:55.39kyronLOL
16:55.51drmessanoWhats really bad.. is that they have NO plans for PSTN connection.. So until then, they're plugging using your old PBX + OLCS
16:56.06drmessanoand eventually, when we're ALL using SIP, you can toss the old one
16:56.12drmessanoSo you STILL need a PSTN PBX
16:57.27kyronMS PBX EULA: "By clicking YES below you agree to Microsoft(c) collecting information on the usage of the system, we won't tell you what we'll do with it, won't tell you we're actually recording all your calls and will definately not state what our future intentions are concerning the data"
16:57.33riddleboxkyron, thats how Avaya got the IP Office
16:57.52kyronLOL...nice plan M$
16:58.00drmessanoIsnt that that the Trixbox EULA?
16:58.05drmessanooh
16:58.13kyronLOL
16:58.25drmessanoYou just switched the names out.. right?
16:58.27drmessanoI get it
16:58.44angryusershould i turn on any option to make * BLF work as specified in RFC 4662 ??
16:59.00drmessanoso app_backdoor is NOT a core asterisk module?
16:59.20angryuser<PROTECTED>
16:59.20kyronkyron@kyron ~ $ wtf is blf
16:59.21kyronGee...  I don't know what blf means...
16:59.32kyronoops...wrong window, sorry guyz
16:59.34riddlebox?
17:00.01kyrondrmessano, LOL, app_backorifice
17:00.13drmessanolol
17:00.22troubledheh
17:00.47drmessanoIf you install Trixbox 2.6 and every call begins with "This call is sponsored by...."  Then you'll know
17:01.20kyronPWahahaha
17:01.20troubledmost likely a MS Office Communications Server 2007 dependency
17:01.37troubledapp_bo that is
17:01.53riddleboxdrmessano, I was talking to someone lastnight who said they were going to install trixbox because asterisk doesnt have good enough documentation, haha
17:02.00kyronso what about the Astra phones...I am a little tempted by them due to their "classic" Nortel T2708 look (which is what my customer has)
17:02.01drmessanoROFL
17:02.21kyronriddlebox, LMAO
17:03.19drmessanoI started off using TB, and it took me some time to realize just how easy it is to do it yourself.. I know that some people consider Fr**PBX a swear word, but you can build 99% of a Trixbox yourself without all the phoning home and crap
17:03.42drmessanoBut some people are just BLIND
17:03.54drmessanoI mean, they really think they NEED it
17:04.43drmessanoand dont dare go in there and get into a war on the forums..   I think they bus in users from Digg to help out
17:05.26riddleboxdrmessano, alot of people want gui's for everything too
17:05.28drmessanoI didnt know Drupal had a fanboy module until then
17:05.37kyronwell, I am presently using tr**box because I don't have time to build an appropriate server. When I finish writing up my damned masters I will build a proper Gentoo box and add * to it and _it_ will be my real PBX ;)
17:06.07riddleboxI compiled asterisk on ubuntu, and it works perfectly
17:06.53drmessanoIt took me.. 3 or 4 days... Mostly nights.. to take docs I found, and document LINE BY LINE how to install Asterisk and FreePBX.. I did a second install in under 2 hours from my docs, and havent looked back
17:07.06kyronriddlebox, I don't doubt that... is the ubuntu kernel built with CONFIG_HZ=1000 ?
17:07.26riddleboxkyron, not sure
17:07.39kyronahHA!...so you still _did_ install Fr**PBX
17:08.09drmessanoI think hand edits AND a GUI have their place..
17:08.20troubledkyron: i gather i want to have 1k for HZ for *?
17:08.26kyronmy intent is to scrap that... ie: I have barely started to use my system and I am already hitting a wall...can't easily configure outgoing trunks depending on originating extension
17:08.28drmessanoFor my PBX here, I use FreePBX.. I have another box I test on, its straight asterisk
17:08.33kyrontroubled, yes
17:08.53troubledkyron: even when im using a hardware/tdm for clocking?
17:09.06drmessanoTo me it depends on the need, and who is going to admin it..
17:09.08kyronhardware tdm?
17:09.14troubledtdm400p
17:09.29kyronoh, I would still say yes
17:10.09drmessanoI've done some Trixbox installs for others and I regret it now.. Theres some crazy crap going on under the hood
17:10.10kyronespecially since chann_sip isn't "proper" yet ;)
17:10.14troubledfor some reason i think debian defaults to like 250. ill have to keep that in mind, thanks
17:11.55*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
17:12.51kyron250 is the default for most distrib and it's quite sound, but for "realtime" apps like *, you're better off with 1k. NOTE: if you have a multicore (not an HT) processor, be aware that CONFIG_HZ is applied to each core...so 1k on a core 2 duo = 2kHz int.
17:13.29troubledno HT here unfortunately
17:13.46DarKnesS_WolFhow can i add my country dialtone / busytone and so on into asterisk to do load it for zaptel ?
17:14.37troubledindications.conf?
17:15.01kyrontroubled, echo "unfortunately" | sed -e 's/un//'
17:15.20troubled:)
17:18.01kyronyou're better off treating an HT processor as a single proc...but the kernel is more better smarter than Microsoft about that :P
17:21.52*** join/#asterisk nybble (n=nybble@about/apple/performa/nybble)
17:25.18angryuserhm, i dont receive any SUBSCRIBE from snom phones, any ideas? worked well with 1.4.0 not with latest
17:26.14troubledanyone know how i can query an iaxy to get its current firmware version? i see firmware updates in trunk@HEAD for it, but no idea if/what ver it is already. and if the updates work ok with 1.2 *
17:31.25J4k3the best thing to do with an HT-equipped processor in 2008 is hit the piece of shit with a hammere
17:31.34J4k3save the environment, smash a P4.
17:32.18troubledi take it you guys are having some problems with HT and *?
17:32.41drmessanoJ4k3: You obviously sound like a mean who doesnt believe in refrigerated cooling systems for PCs
17:32.42*** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo)
17:33.09drmessanoman
17:33.09drmessanowow
17:33.16kyronLOL, J4k3 I wouldn't go that far
17:33.31kyronecologically speaking, keeping a machin in the works is better than replacing it...
17:33.37drmessanoha
17:33.41kyron* runs fine on a P3
17:34.24DarKnesS_WolFtroubled: okay thx where i can get them :P the tone it self
17:34.36drmessanoNetworld World, who spams me three times a day wanting me to change to the online edition, now wants me to "go green" and use the online edition instead of print
17:34.53troubledDarKnesS_WolF: get what now? ring indications?
17:35.00angryuseri have found it! ;)
17:37.39angryuserto share knowledge in version 1.4.0 option subscribecontext ="" had no effect on snoms lastest firmware and latest version of * corrects that, i had a wrong subscribecontext option set
17:38.04J4k3kyron: yeah, but most P4's consume 4-6x the power of a high end P3, and don't deliver the pereformance except in very specific instances
17:38.13*** join/#asterisk batphone (n=batphone@70.124.59.254)
17:38.31batphonewhat could cause there to be no sound from voicemail greetings?
17:38.40J4k3the P4 was a joke the day it was released in all its 1.6ghz glory, and it sucked until the day the core2 was released :P
17:39.14angryuserbatphone: no core sound files?
17:39.19kyronGuyz, calm down, let's go PS3 ;)
17:39.20batphoneroot@iev:2113> ls -l /dev/dsp
17:39.20batphonecrw-rw----  1 root audio 14, 3 Jul 26  2005 /dev/dsp
17:39.23batphonemaybe that..
17:39.30tzafrir_homeJ4k3, the P4 had a problem initially because code optimized for P2/P3 was badly optimized for the P4
17:40.41*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
17:40.47J4k3tzafrir_home: yeah, and optimizing for the P4 causes code to suck... even on the P4 (see the gcc nocona performance issues) :)
17:41.01kyrontzafrir_home, which comes to the point when I tell people CentOS is a Microsoft of the Linux world... an OS built for the Pentium Pro :P
17:41.18kyronuse ICC ;)
17:41.42drmessanoCentOS is great
17:41.49PakiPenguincentos is awesome
17:42.03TJNIIbatphone: No sound over your phones, or on the console?
17:42.08kyronO_o....
17:42.19angryusernever used CentOS, dont know why, used to debian, why is it great.
17:42.21angryuser?
17:42.35PakiPenguincentos is just awesome , very easy to setup , very stable
17:42.41drmessanokyron, you're using Trixbox, which is built on CentOS.. should you be throwing stones?
17:42.46DarKnesS_WolFtroubled: 425 Hz modulated with 50 Hz
17:42.49batphoneTJNII: no sound from sound files
17:42.56troubledand probably the most geared to * out of the box :)
17:43.00DarKnesS_WolFtroubled: now i have to understand how can i put this in indections.conf ;-)
17:43.16DarKnesS_WolFPakiPenguin: hello dude ! i don't see u that much in shellshark anymore
17:43.21TJNIIbatphone: yea, over your phones, or on the console?
17:43.25kyrondrmessano, I intend to scrap this POS for a GEntoo build with manual built * as soon as I get a chance (and read the * manual _first_ :P)
17:43.39troubledDarKnesS_WolF: im not sure how zap driver grabs that info. might be hardcoded in the chan.c or .h files somewhere
17:43.40kyrondrmessano, I basically used Trix for a proof of concept
17:43.46kyronnot for a serious implementation
17:43.50batphoneTJNII: phones i guess, i dont know what you mean by sound over the console, im doing this remotely
17:44.01drmessanokyron: thats what they all say :)
17:44.09PakiPenguin:) hey DarKnesS_WolF
17:44.13troubledDarKnesS_WolF: i suppose you can just put in a new country code with the changes and try restart though
17:44.16PakiPenguinyeah been busy with work
17:44.17DarKnesS_WolFtroubled: zonedetec.c as i think or so i read it somewhere but i still can add it into indecations.conf and see
17:44.25troublednod
17:44.31DarKnesS_WolFPakiPenguin: cool what new toys u did ?
17:44.31TJNIIbatphone: Okay, If it is over the phones then it wouldn't be a permission problem on /dev/dsp.
17:44.40PakiPenguinyeah nokia r4
17:44.44drmessano"This is my other car"  "This is my other PBX".. same difference
17:45.10batphoneTJNII: for example, when i call the system and the call times out and gets routed to voicemail, the voicemail announcement wont play
17:45.20TJNIIbatphone: Does the console output say anything about missing file?
17:45.33batphoneTJNII: the file is present and the console states the file is being played
17:45.43batphoneTJNII: -- Playing '/var/spool/asterisk/voicemail/default/2113/temp' (language 'en')
17:45.54kyrondrmessano, you underestimate my dislike of CentOS... and FreePBX doesn't let me configure outbound route selection from source out of the box...frustrating
17:46.15batphoneTJNII: im wondering if the encoding got messed up
17:46.26TJNIIbatphone: Does the greeting play?
17:46.41PakiPenguinDarKnesS_WolF, what are you upto?
17:46.49batphoneTJNII: no
17:46.51troubledbatphone: can you play that wav file and check if it recorded properly, and not just empty noise?
17:46.55batphonetemp.wav:    RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
17:47.08batphonetroubled: the autoattendant does not play
17:47.16batphonetroubled: TJNII: lemme download those files and see wtf
17:47.37batphonetroubled: TJNII: yeah, no greeting, no sound files play whatsoever, i dont think its the files
17:47.52TJNIIbatphone: If nothing plays you are probably right, unfortunately
17:49.40TJNIIbatphone: I don't know enough about the subsystem that converts formats and links channels to really help you from here.  Sorry.
17:50.31batphoneTJNII: troubled: the files play on my machine, sound good too
17:50.42troubledbatphone: curious, the path for "-- Playing" didnt have a file extension, i assume that implies that it will use the format for the codec used for that phone type. is it possible its looking for a .gsm?
17:50.59troubledtry force the phone to .wav only?
17:51.16batphonetroubled: i think there is an order in which asterisk looks for files to play, but i'll try explicitly adding it
17:51.22troubledi was under the impression the voicemail system created all the files automagically
17:51.23TJNIItroubled: It should try to use the best codec, and convert if necessairy.
17:51.30troubledah
17:52.00TJNIIbatphone: Did you install froum source?
17:52.16batphoneTJNII: this is an old box, gentoo, running asterisk 1.2.7.1
17:52.21troubledTJNII: so it should recognize the .gsm isnt there for example, and convert the .wav on the spot?
17:52.34TJNIItroubled: To the best of my knowledge, yes
17:52.41batphoneTJNII: troubled: voicemail.conf says format=wav so it should be finding it
17:53.41TJNIIbatphone: Since nothing is playing the problem is probably below the settings in voicemail.conf
17:54.16TJNIIbatphone: try exten => 100,1,Playback(hello-world) and see if it plays
17:55.28batphoneTJNII: trying
17:55.40batphoneTJNII: no go!
17:56.07TJNIIbatphone: And you have the verbosity turned up on the console, right?
17:56.26batphoneTJNII: cranking
17:57.01batphone<PROTECTED>
17:57.04batphoneall it says
17:57.10TJNIIhmmm
17:57.14batphonebesides the call execution stuff
17:57.28TJNIIWell, the problem isn't your voicemail settings, we've proved that much.
17:58.01batphonei built this thing 2 years ago. they let some dude get into it and now everything is messed up
17:58.19batphone100 phones, 4 remote offices.. ugh. i worked my ass off on this thing
17:58.49TJNIIbatphone: Well, it is gentoo.  Can you jusr to an emerge asterisk and see if it fixes itself?
17:59.00batphoneTJNII: i'd like to but the dial plan is massive and i
17:59.02TJNIIs/jusr to/just do/g
17:59.04batphoneid hate to roll this back
17:59.28TJNIIbatphone: cp /etc/asterisk /root/asterisk.backup.011907 -rv
17:59.44TJNIIWill backup all your configs
18:00.24batphonedone
18:00.32batphonerebuilding
18:00.41*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:01.28TJNIIbatphone: It just seems like he broke something pretty low level, so a re-emerge is worth a shot.
18:01.37batphoneTJNII: agreed
18:01.52batphoneTJNII: i dont see any error messages for the hardware anywhere
18:01.58tzafrir_homeTJNII, cp -a instead of just cp -r . Preserve dates, premissions, etc.
18:02.13TJNIItzafrir_home: noted
18:02.31*** join/#asterisk atisss (n=atisss@193.238.212.171)
18:02.51TJNIItzafrir_home: That's the command I use whan I'm afraid a update is going to hork something.  For an honest backup I do a tar -cvp
18:03.27tzafrir_homeTJNII, put /etc/asterisk under version control :-)
18:04.09TJNIItzafrir_home: Heh.  One of the many things I *should* do
18:04.29batphoneTJNII: rebuilt, still no soudn
18:04.34TJNIItzafrir_home: Actually, all of /etc under version control would probably not be a bad idea
18:05.34TJNIIbatphone: Do normal calls work OK?
18:06.14batphoneyes
18:07.17*** join/#asterisk ManxPower (n=manxpowe@209.16.72.139)
18:08.51TJNIIbatphone: I really don't know enough about how * links sound files to the phones to help from here.  It looks like that part of * is messed up.
18:08.56*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
18:09.13*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
18:09.36DarKnesS_WolFPakiPenguin: nothing i have my own * at my place now :-) with fax detection and will upgrade it to 1.6 and test :-)
18:09.49TJNIIbatphone: If I was in your shoes I would try an emerge --sync and an emerge -uDNvp world and look for sound conversion programs like sox that it wants to update.
18:09.57PakiPenguinah i see
18:09.58PakiPenguinnice nice
18:10.29DarKnesS_WolFPakiPenguin: also now trying to put the indecations of my country :-)
18:10.38PakiPenguinah i see
18:10.39ManxPoweryou realize that packaged Asterisk is not support by anyone but the packager, right?
18:11.00troubledDarKnesS_WolF: what country anyways?
18:11.07DarKnesS_WolFegypt
18:11.14DarKnesS_WolFtroubled: u ?
18:11.24troubledah. not already in there i assume? .ca here
18:11.39DarKnesS_WolFtroubled: no not htere .. and i think we are using EU things
18:11.59troubledya, "us" seems to work for me here
18:12.00DarKnesS_WolFbut i'm trying to understand how to add indections. alittle bit cool to have same tone on ur SIP phone
18:12.09DarKnesS_WolFactually i'm using US too
18:12.19ManxPowerUnfortunatly inidications are a mess in voip and Asterisk
18:12.23DarKnesS_WolFbut egypt strange country every PSTN section has it is own servers .
18:12.32DarKnesS_WolFManxPower: long time no seen dude :-) and why is that ?
18:13.11ManxPowerzaptel seems to have manh it's own inidcations hard coded.  Each SIP phone has it's OWN inidcations set on the phone outside of Asterisk, then you have out of band indications which are sent before the call is answered and inband inidcations that are sent after the call is answered.
18:14.47ManxPowerI've only seen /etc/asterisk/inidcations.conf change inband indications and Playtones, nothing else.
18:17.29batphonei am so stuck...
18:17.31DarKnesS_WolFManxPower: ok solution ?
18:17.36DarKnesS_WolFbatphone: in what?
18:17.40troubledi assume the FXO modules are hardcoded cause of FCC/CRTC reasons?
18:18.09ManxPowerDarKnesS_WolF: change it everywonere
18:18.14batphoneDarKnesS_WolF: sound files dont produce sound
18:20.44tasteriskCan anyone recommend a cheap asterisk hosting provider?
18:21.34batphoneDarKnesS_WolF: for example, auto attendants dont work, voicemail doesnt work, anything sound related simply does not work
18:21.52batphoneDarKnesS_WolF: there is no amount of verbosity i can use in logging to produce any useful output
18:21.53DarKnesS_WolFtasterisk: ur own machines ;-)
18:22.03DarKnesS_WolFbatphone: codecs ?
18:22.07troubledbatphone: what exactly is the sound output device you are using again?
18:22.16batphonetroubled: /dev/dsp
18:22.24batphoneDarKnesS_WolF: what about them?
18:22.31batphoneDarKnesS_WolF: not sure where to begin there
18:22.36troubledbatphone: and what groups in asterisk a member of again?
18:22.41DarKnesS_WolFbatphone: ur not using a phone to listen to the voicemail ?
18:22.49batphoneroot@iev:asterisk> groups asterisk
18:22.49batphoneaudio dialout asterisk
18:23.01DarKnesS_WolFbatphone: 1st thing i would try is to run asterisk as root
18:23.11DarKnesS_WolFusing init.d script or even asterisk -vvvvvvc
18:23.23DarKnesS_WolFthen get a softphone or ip phone sip/iax register it to asterisk
18:23.26DarKnesS_WolFand call the voicemail
18:23.55DarKnesS_WolFwatch the CLI and see what u see whe nu dial ur voicemailmain exntesion and watch.
18:24.24batphoneDarKnesS_WolF: running as root didnt work
18:24.31DarKnesS_WolFbatphone: errors ?
18:24.48batphoneDarKnesS_WolF: none, Jan 19 13:25:02 DEBUG[22361]: channel.c:1711 ast_settimeout: Scheduling timer at 160 sample intervals
18:24.57batphonethat is the only interesting log entry
18:25.11batphonebesides the     -- Playing 'hello-world' (language 'en')
18:25.43troubledbatphone: does any sound come out the /dev/dsp?
18:25.46troubledfor anything that is
18:26.06batphonetroubled: dont know, im doing this remotely so i cant really pop in some headphones and jam
18:26.18batphoneanyone know of a way to test that remotely?
18:26.48troubledhow do you know that its not working then?
18:27.32batphonetroubled: im not sure i follow you.
18:27.35SteveTotarolol
18:27.46troubled:(
18:27.48batphonetroubled: when you call the box, no sound files play
18:28.07batphonecall -> ring -> voicemail -> zero sound
18:28.09troubledbatphone: but you are expecting them to output to /dev/dsp, but you just told me you cant hear it
18:28.26troubledthen the output isnt the dsp, its your phone, right?
18:28.42batphonetroubled: ok i guess i misunderstood you, yes the output is my phone's dsp
18:28.53troubledor are you using a softphone that uses /dev/dsp on your computer you are testing from?
18:28.54batphonetroubled: but isnt asterisk using /dev/dsp to process sound...
18:28.54DarKnesS_WolFbatphone: dude wakeup :-s
18:29.12batphonetroubled: my cell, anyones cell, from anywhere in the world
18:29.19DarKnesS_WolFbatphone: do u have a softphone ?
18:29.24batphoneDarKnesS_WolF: on
18:29.27batphoneDarKnesS_WolF: no
18:29.34SteveTotaroget one
18:29.44DarKnesS_WolFbatphone: ur client windows or liunux ?
18:29.55batphoneDarKnesS_WolF: what client? im running linux.
18:30.00DarKnesS_WolFget ziper for windows or linux create IAX2 client
18:30.05DarKnesS_WolFin ur asterisk in iax.conf
18:30.19batphoneDarKnesS_WolF: so we are just trying to see if this is a sip related issue?
18:30.27ManxPowerbatphone: rmmod ztdummy and try it
18:30.37batphoneztdummy not loaded
18:30.42DarKnesS_WolFand then register the phone to it and iax show users in asterisk CLI u should see it is reigsted and then dial ur VoiceMailMain aapplication.
18:30.46batphonei have an fxs card in there...
18:30.50ManxPowerare you sure?  It can load automatically.
18:30.51*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
18:30.55ManxPowereven if you have a card
18:31.02DarKnesS_WolFbatphone: nop i just want to give u a good testing evn. what ur doing will never get u to the source of the probelem
18:31.13DarKnesS_WolFbatphone: and u have a phone connected there !?
18:31.14ManxPoweryou know if you have a card installed but not configured it can also cause loss of sound.
18:31.21DarKnesS_WolFu said it is remolty ! how can u pick up the phone there ?
18:31.39batphoneDarKnesS_WolF: its not voice audio, its sound files on the server that dont play.
18:31.54batphoneDarKnesS_WolF: i should be able to call the system and have hello-world play into my ear, right?
18:33.09ManxPowerbatphone: yes you should be able to.  does the CLI show anything useful?
18:33.31batphoneManxPower: no
18:33.37batphoneManxPower: check this, got sound now...
18:33.41ManxPowerI assume by "call the system" you mean "pick up a phone attached to the FXS card and dial an extension that does a Playback().
18:33.47ManxPowerbatphone: what did you do?
18:33.51batphoneManxPower: i removed all zaptel modules and started asterisk w/o zaptel
18:34.01ManxPowerbatphone: It's nice to be right.
18:34.29batphoneManxPower: gotta fix this though. starting in on zaptel....
18:34.30DarKnesS_WolFbatphone: yes but ow are u calling ?
18:34.36puzzledhow do you dial an international number from the US? is that 011<country><area><number>?
18:35.03SteveTotarodepends on the country code
18:35.05DarKnesS_WolFManxPower: but why ztdummy did this !?
18:35.06batphoneManxPower: bad card...
18:35.17ManxPowerbatphone: As I said.  ztdummy can load automatically and if the kernel RTC is screwed up you can get no audio.  Also if a zaptel card is installed and the drivers are loaded, but the card is not configured that can also cause the same issue.
18:35.18batphoneDarKnesS_WolF: ztdummy had nothing to do with it
18:35.21DarKnesS_WolFpuzzled: and ur provider some using 00
18:35.27DarKnesS_WolFso u unload zaptel ?
18:35.34batphoneDarKnesS_WolF: yes
18:35.38batphoneDarKnesS_WolF: stopped it completely
18:35.40DarKnesS_WolFbatphone: ManxPower dosn't make sense
18:35.49ManxPowerDarKnesS_WolF: Why not?
18:36.17ManxPowerAsterisk will try to use the card to get audio sync timing and if the card is not generating interrupts (because it's not configured) asterisk will be silent.
18:36.35SteveTotaroi have never experience that
18:36.38ManxPowerI've only PERSONALLY seen this on T-1 zaptel cards.
18:36.46ManxPowerI don't know if it is possible on analog zaptel cards.
18:36.47batphoneDarKnesS_WolF: he is absolutely right. Zaptel is, well, zapped in this box
18:36.56ManxPowerSteveTotaro: Others have seen the issue as well.
18:37.00batphonei think its the card, its throwing channel errors now
18:37.20SteveTotaroi use all sangoma though
18:37.23ManxPowerbatphone: nothing quite like being vague about error messages.
18:37.30batphoneManxPower: theres alot, digging
18:37.36ManxPowerSteveTotaro: I do now.
18:38.51puzzledanyone know what could be causing this error message: Failed to authenticate on INVITE to '"Test 1001" <sip:<number>@213.84.38.78>;tag=as14dce4da'
18:39.05puzzledI have canreinvite=no everywhere
18:39.24ManxPowerpuzzled: all calls have at least one invite
18:40.20puzzledManxPower: you mean one reinvite to the remote proxy or from the local phone to the local asterisk box?
18:40.27puzzleds/reinvite/invite
18:41.20DarKnesS_WolFbatphone: ok load zaptel again and then do
18:41.26DarKnesS_WolFztcfg -vvvvvvvvv
18:41.29DarKnesS_WolFon ur shell
18:41.47*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
18:41.47*** mode/#asterisk [+o russellb] by ChanServ
18:42.09DarKnesS_WolFrussellb: hello dude :) welcome back :-D
18:42.25russellbthanks
18:44.06DarKnesS_WolFrussellb: any idea how to add indections on indecations.conf ? i only have this Frequency: 425 Hz modulated with 50 Hz
18:45.43*** join/#asterisk UnixDog (n=unixdog@adsl-69-234-189-103.dsl.irvnca.pacbell.net)
18:47.32russellbum, is it not documented?
18:47.53troubledheh
18:48.23troubledsome say irc is a form of documentation you know ;)
18:48.42russellbDarKnesS_WolF: it would just be ... mytone = 425+50
18:48.50troubledSDP, streaming documentation protocol
18:49.16*** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net)
18:50.15russellb~sdp
18:50.33russellbdarn
18:50.35russellb:)
18:51.00mihinomenestquick question, management wants me to configure * so that if a caller's in the queue, they can press 3 for example, and get out of the queue and into voicemail.  is this possible?
18:51.03ManxPowerrussellb: What does /etc/asterisk/inidcations.conf control.   Not SIP tones, of course.  Does it control zaptel fxs tones?
18:51.12ManxPowermihinomenest: yes.
18:51.15troubled~sdp is <reply> IRC is known to be a form of the Streaming Documentation Protocol.
18:51.16jbotokay, troubled
18:51.24troubled;)
18:51.52mihinomenestManxPower: ok, I presume that's all on voip-info somewhere?
18:51.54DarKnesS_WolFrussellb: ok what about busy tone and all this stuff?
18:51.54J4k3haha
18:52.20ManxPowermihinomenest: Examples should be there.  "show application queue" and queues.conf should help you there.
18:52.27*** join/#asterisk [koss] (i=koss@adsl-75-36-15-24.dsl.bcvloh.sbcglobal.net)
18:52.32mihinomenestgreat.  thanks.
18:53.02*** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177562101.dsl.bell.ca)
18:53.32russellbManxPower: it controls any tones generated by Asterisk.  Zaptel has its own set of tones it generates.  But, SIP does actually use this in some cases (where tones have to be generated in-band, like in-band ringing, progressinband=yes, or depending on what SIP response we get)
18:54.14russellbyou can also manually play these tones from the dialplan ... i don't remember the application name(s)
18:55.00DarKnesS_WolFrussellb: mmmmm not sure i'll go get some sleep i was on my weekend and i had to work for asterisk client :P so i'll go home sleep and tomorrow will digg it
18:56.24russellbheh, ok
18:57.34WilliamKhey russellb, sorry I didn't see the msg from Danny until you were gone yesterday... however, from my viewpoint and probably others... Digium may want to protect the Digium name, while the "Asterisk" software itself is community based... although from Danny's standpoint he/others inside Digium may want to work with others to create a "controlled" list of those known selling Digium hardware just for monitoring
18:57.37WilliamKpurposes
18:57.49troubledrussellb: hey, while i got you here, do you know anything about iaxy firmware updates? curious how i can tell what i have in mine now, and if i should apply the iaxy.bin for trunk@HEAD even though im using 1.2 in debian
18:59.17SteveTotarothe whole thing is quite silly with the trademark stuff
18:59.38troubleddid i miss a memo?
19:00.01SteveTotaroyes, from the CEO himself apparently
19:00.04WilliamKSteve, I agree, but I also understand what Danny intended
19:00.24SteveTotaroComing from a totally different paradigm maybe
19:00.36WilliamKI don't think he intended on shooting himself in both feet though but slashing his revenue
19:01.01troubledbeing a "Danny" myself, im finding this conversation slightly confusing ;)
19:01.07mihinomenestOk, that was stupid easy.
19:01.12mihinomenestthanks ManxPower.
19:01.20WilliamKultamately Digium makes the hardware, to stop others from reselling it no matter how much you dislike them causes a damage to revenue
19:01.48SteveTotarowell I will tell you one thing, Sangoma would never do this
19:02.17SteveTotarothey don't even require any kind of "ecosystem"
19:02.19WilliamKnow on the other hand you catch someone counterfiting your patents/designs... I would say be aggresive in stopping them
19:02.33WilliamKbut if it's based on OpenSource then it needs to not be filtered in any way
19:02.48WilliamKwhich * is considered OpenSource / Ecosystem
19:02.51russellbtroubled: i don't think the iaxy firmware has been changed in ages.  the one in 1.2 should be the same as trunk.
19:03.08*** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177562101.dsl.bell.ca)
19:03.53*** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com)
19:04.09*** join/#asterisk Winkie (n=urmom@87-194-109-4.bethere.co.uk)
19:04.20SteveTotarowhen i reference the "ecosystem" i am talking about Digium's stance for being an approved vendor
19:04.27*** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo)
19:04.36SteveTotaroyou must agree to only sell Digium hardware
19:04.44WilliamKah that
19:04.50WilliamKthat itself is not good
19:05.24SteveTotaroi wonder about companies like voipsupply
19:05.25WilliamKthat defies the free enterprise system goals and market competiveness
19:05.40WilliamKabout ?
19:05.47SteveTotaroyes, it is moving to a proprietary system
19:05.55troubledrussellb: oddly enough, /usr/share/asterisk/firmware/iax/ is an empty dir in etch
19:06.00SteveTotaroexactly what asterisk was *not* supposed to be
19:06.37troubledrussellb: i gather it should be more than safe to pop that iaxy.bin i have right into that dir, reload asterisk and reconnect the iaxy and *poof*?
19:06.57Nivexthat doesn't make it proprietary.  you have access to the code, so if you want to, you can create your own hardware that is Asterisk compatible
19:07.21WilliamKwell an interesting thing is... I remember Mark Spencer by face/name, but until the other day I had never heard of Danny... which has me wondering about how much Danny is involved in OpenSource
19:07.29SteveTotaroyes but if you are a vendor using adwords you would have had a problem
19:07.56WilliamKIf the environment for open source doesn't prosper and investors take over then $ is the only value they see, thus defying the OpenSource commitment
19:08.01SteveTotaroDanny was Mark's boss when he was an intern at Adtran
19:08.15WilliamKwhich is my whole concern for the Sun takeover of MySQL
19:08.52SteveTotarobut that paradigm doesn't fit with opensource
19:10.15WilliamKI view OpenSource as everyone is freely able to make changes, post changes/fixes, and drivers for hardware creations should also be bundled if the vendor of the hardware chooses... thus Sangoma and others would be transparent when installing their boards
19:10.30WilliamKhowever, it may hurt Digium in the long run by open marketshare
19:10.42russellbtroubled: yeah, you can stick it in there.  it's not in debian because it's not open source.
19:10.56SteveTotarothat is where build quality comes in
19:11.27SteveTotarowhoever makes the best hardware should get the market share
19:11.34WilliamKI agree that there should be others checking for build quality, but they should be unbiased as to one vendor's items over another
19:12.02SteveTotaroactually, i am looking at doing that myself
19:12.15WilliamKfrom what I've seen of the SVN builds lately especially when compiling... it's been VERY bad on CentOS at least...
19:12.33WilliamKwhich tells me little qual checks are being done before/after commit against multiple platforms
19:13.02SteveTotaroi build large scale systems and only feel comfortable with 1.2.x
19:13.11troubledrussellb: ahh, good to know. curious though, is this a perm flash? or just something temprary that needs to stay around across reboots? and what are odds of bricking my iaxy?
19:13.35WilliamKI'm kinda surprised at this promotion to 1.6 while I think 1.4 has a long ways to go yet
19:13.37russellbtroubled: i wouldn't worry about bricking it.  also, it's a permanent thing, doesn't need to be there across reboots.
19:13.47troubled:)
19:13.50SteveTotaroi think it (1.2) should be spooned (a nicer way to say forked) and features bugfixes added instead of just security fixes
19:14.01troubledany idea if i will see something in the logs about sucess?
19:14.12russellbtroubled: no clue, but don't think so
19:14.30troubledlets find out shall we :)
19:14.49SteveTotaroi think the iaxy does log something about firmware when it registers
19:14.53WilliamKSteve, I somewhat agree but understand it's double the time for developers to go backwards and check
19:15.06WilliamKunless someone wants to do build testing for older platforms
19:15.38SteveTotarothat is why i say fork, i know many people who also do large scale deployments that won't touch 1.4
19:16.10ManxPowerI have high hopes for 1.6.  Chances are we will skip 1.4 and go directly from 1.2 to 1.4
19:17.03WilliamKI'm just intrigued by features that users want and can't have yet which would promote marketshare of *
19:17.06WilliamKjust seems odd
19:17.14fileit's a full time job... unless there was a large amount of individuals
19:17.18troubledhmm, nothing showed in the logs
19:17.34ManxPowerWilliamK: Features such as....?
19:17.37WilliamKfile, I understand... teach me programming :)
19:17.49filesilly finite amount of time
19:18.01troubledheh
19:18.17DarKnesS_WolFno way to have an opensource integration with skype and asterisk soon ?
19:18.21WilliamKManx, TLS has been one for a long time, Fax that works, etc...
19:18.22SteveTotarorussell definitely does not like the real world truth
19:18.41WilliamKI saw the changelog for 1.6, looks better but I still think the hammer missed the head on some things
19:18.46ManxPowerUsers don't even understand TLS.  And fax seems to work for us just fine.
19:18.47SteveTotarothe only feature i see a need for is whisper paging
19:19.10DarKnesS_WolFWilliamK: it is amazing i'm downloading 1.6 for my laptop asterisk and my home asterisk ;-) moving already
19:19.20tzangerwtf is whisper paging?  You're on the phone and the system plays to your handset "calling Dr. Potts" ?
19:19.25DarKnesS_WolFSteveTotaro: what is whisper paging?
19:19.26WilliamKManx, I've tried constantly and cannot force it to work
19:19.46SteveTotarowhisper paging is good for coaching agents in call centers
19:20.05tzangerSteveTotaro: that's already there in 1.4, I ran across it the other day in my own * system
19:20.07ManxPowerWilliamK: I guess it depends on which of the 7 million ways Asterisk can "support fax" that you are trying to use.
19:20.09SteveTotaroi do large scale call center deployements, spoke at astricon about it
19:20.26drmessanoFax over SIP or over PSTN?
19:20.32WilliamKjust passthrough would be nice
19:20.45SteveTotarobut i will not deploy 1.4 in a high volume call center
19:20.53WilliamKwent and got a T.38 capable ata after the dang error messages and still can't make it work
19:21.00drmessanoFax and SIP don't mix well, that's not Asterisks fault
19:21.03ManxPowerWilliamK: Ah.  We don't actually do any of that.  We use spandap + rxfax + custom script to receive faxes on the person's DID.
19:21.11tzangerwhere the hell did I see that
19:21.23ManxPowerThe actual fax machines are on a channel bank on a CT1 totally outside of Asterisk
19:21.51SteveTotarochannel bank is the best way to go
19:21.55SteveTotarofor faxes
19:21.59DarKnesS_WolFWilliamK: i have this setup and i think x86 has it too ,,, for me PSNT ----> Digium card -----> asterisk with nvfaxdetect and nvbackgrounddetect -----> to iaxmode same machine -------> hylafax ------> pdf -----> email
19:22.01tzangeragreed, that's what I've done
19:22.04ManxPowerrock solid, no issues.
19:22.09SteveTotaroor iaxmodem and hylafax via crossover cable
19:22.48ManxPowerMost of our users use DID based faxing, but if they have problems with that they can use the PSTN based fax machine.
19:23.00WilliamKI understand the timing issues, and then I see companies making like AsterFax, etc...
19:23.18tzangerahh, it was chanspy
19:23.22troubledDarKnesS_WolF: so asterisk just detects the fax and can be setup to send to userland? or can it accept and store it built in?
19:23.28tzangerSteveTotaro: chanspy lets you whisper... what's a whisper page though?
19:23.34SteveTotarochanspy only allows you to listen
19:23.54WilliamKcurrently the way I've been trying is CT1 --- >Asterisk w/Digium card---- > Linksys 2102--- Fax
19:23.57tzangerSteveTotaro: Listen to a channel, and optionally whisper into it
19:24.33SteveTotarowhisper allows say a supervisor to coach an agent in training without the customer hearing
19:24.34WilliamKwon't work at all inbound
19:24.34WilliamKworks sometimes outbound
19:24.34tzangercore show version
19:24.34tzangerAsterisk 1.4.13 built by root @ gromit on a i686 running Linux on 2007-10-29 17:26:29 UTC
19:24.34tzanger1.4.13 apparently has whisper mode on chanspy
19:24.37ManxPowerWilliamK: Why so complicated?
19:24.37SteveTotaroprobably, again, i don't use 1.4
19:24.43tzangerSteveTotaro: ah
19:24.47tzangermissed that part
19:24.49WilliamKManx, that's complicated?
19:24.54ManxPowerSteveTotaro: I think we are the last two 1.2 users on the planet.
19:24.54tzangeryou and ManxPower are in the same boat :-)
19:25.04ManxPowerWilliamK: using an ATA is what makes it complicated.
19:25.06WilliamKthat's (2) boxes and a fax machine
19:25.17SteveTotarothe call centers i setup would lose tens of thousands of dollars for a single hour of downtime
19:25.18ManxPowerYou KNOW FaxOverVoiceOverIP is not expected to be reliable.
19:25.43*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
19:25.43*** mode/#asterisk [+o lmadsen] by ChanServ
19:26.00WilliamKwhen you're sitting on the same network it should be reliable, I understand if you're on a cable modem
19:26.30WilliamKwhat is astounding is that I can fax reliably off my Linksys ATA through deltathree/Verizon and have it reliable everytime
19:26.48SteveTotarot38
19:27.03ManxPowerWilliamK: It does not work reliable for you.  Find a way to make it reliable.
19:27.10WilliamKand the 2102 supports T.38 which * is lacking in a perfect model
19:27.33ManxPowerThese are not toys we are working with, these are people's phones and faxes we are working with.
19:27.47SteveTotaromanx, i know quite a few using 1.2
19:27.51WilliamKManx, if I knew programming as much as I know business methods and network design then it'd be fixed
19:27.54WilliamK:)
19:28.02troubledManxPower: not true, im 1.2  :)
19:28.11WilliamKthus file's job to teach me programming!
19:28.23filejust because you know programming doesn't mean you can write a fax modem and T38 support
19:28.24ManxPowerWilliamK: You already know what works.  But you are not using it.
19:28.29fileit's complex ^_^
19:28.57ManxPowerOr more correctly, you know what does NOT work well and you are not trying other ways.
19:29.24WilliamKI've tried g711 passthrough through *, and also the limited T.38 and none work
19:29.39WilliamKu got other methods that involve utilizing a fax machine?
19:29.48filenot all T38 implementations are equal, and not all networks are equal
19:30.17ManxPowerWilliamK: how about methods that do not involve sending the fax over a data network?
19:30.25WilliamKfile, I'm just saying that network which is under one control and has plenty of capacity should work
19:30.31SteveTotarobingo Manx
19:30.42ManxPowerFXS port on Asterisk or just not run the fax machine via Asterisk.
19:30.48SteveTotaroany latency will affect faxing
19:30.52WilliamKManx, but then what's the point of T.38 or g711 passthrough
19:31.04ManxPowerWilliamK: and I'm saying you are living in a fantasy world with smurfs and unicorns.
19:31.17WilliamKkinda makes no point to develop and continue to innovate
19:31.19SteveTotarolol
19:31.34ManxPowerWilliamK: the point of T.38 is that it is such an immature "standard" that it is no ready for real world corporate mission critical use.
19:31.39SteveTotaroT.38 is just not where it needs to be in asterisk
19:31.39drmessanoCan a unicorn deliver my faxes? :)))
19:31.40WilliamKManx, I live in the human network, thus in your mind :)
19:31.49fileSteveTotaro: quite.
19:32.15SteveTotaroso go with a channel bank
19:32.16ManxPowerSo as long as you have this unhealthy fetish for T.38 your life is not going to be a happy one.
19:32.46drmessanoI tried a T.38 using a provider that "Supports T.38" and a SPA-3102 directly peered to it.. and it didnt work worth a flip
19:32.50SteveTotaroor go with iaxmodem + hylafax on a separate machine connected via crossover
19:32.55WilliamKI take g711 passhthrough too if you can make it work
19:32.58ManxPowerYes T.38 sounds cool.  Yes T.38 could be very useful, Yes T.38 can solve many problems.  But the tech is too immature to do that TODAY.
19:33.07WilliamKwhat's really amusing is a modem can work
19:33.10WilliamKjust not a fax machine
19:33.25SteveTotaroset the fax to 9600 baud
19:33.32SteveTotaroit will probably work then
19:33.44SteveTotarowork *very* slowly
19:34.07drmessanoT.38 will be mature around the time everyone realizes even a $150 HP All in One with a built-in NIC can do scan to email
19:34.20SteveTotarothere you go
19:34.56ManxPowerSome of us know what is the most reliable way to do faxes with Asterisk -- the answer is to not do faxing via Asterisk.
19:35.06SteveTotaroand for real high demand faxing there are some panafax machines that can scan and send 1,000 pages plus
19:35.25DavieyWhy people still need to fax is beyond me
19:35.28ManxPowerAnd when your users start screaming at you that they lost a million dollar contract because you can't make the fax machine work well -- you might start to feel the same way.
19:35.30drmessanoExactly
19:35.51drmessanoId rather scan to a PDF than send a shitty fax
19:35.59SteveTotaroyes, contracts are legal (at least in the US) signed and faxed
19:36.20ManxPowerDaviey: My customers use it mostly for mortgage papers for houses and leases for commercial buildings.
19:36.47drmessanoManxPower: Point made.. if it needs to go to a county or city office, they barely have email
19:36.48SteveTotarosame sector i see it
19:37.00drmessanolol
19:37.02my007msand Doc abut connect 2 asterisk box over SIP trunk
19:37.21my007msi don't need to use IAX
19:37.26drmessanoThere is no such thing as a SIP trunk
19:37.34ManxPowerBTW, I am currently accepting new customers. 8-)
19:37.40SteveTotaroloan document packages can be 50 pages or more
19:37.57DavieyManxPower: fair enough.. but email to fax is a suitable replacement IMO
19:37.57ManxPower~trunk
19:38.07jbot[trunk] is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term.
19:38.07drmessanoIt sure is
19:38.13WilliamKSteve, yeah I know... I hated signing the paperwork for my house
19:38.16*** join/#asterisk endre (i=me2@urbnet.hu)
19:38.31endreMw3: hol?
19:38.40WilliamKclosing person at title company told me about 2-4hrs of paperwork signing
19:38.44ManxPowerDaviey: The last time we looked into e-mail to fax all my users wanted to send MS-Word docs via the system and that didn't work well.
19:38.58SteveTotaronot sure if scanned and emailed contracts are legal
19:39.09ManxPowerI guess we should look at the system again as many of our users now use stuff that can export to PDF
19:39.57SteveTotaroi was a mortgage broker for about six months
19:39.57ManxPowerSteveTotaro: How did you get your soul back after you quit?
19:39.57DavieyManxPower: yeah. OpenXML would be an issue.. standard .Doc should be fine
19:39.57SteveTotaroi quit because i did not sell my soul
19:40.07SteveTotaroi only did loans that made sense
19:40.09*** join/#asterisk Docfxit (n=Docfxit@ip-64-32-143-214.lax.megapath.net)
19:40.18SteveTotarothis was the prime time for sub prime
19:40.23DavieySteveTotaro: whats the difference between scanned doc' emailed and faxed and a fax?  I mean, a fax is just a scanner
19:40.30SteveTotaroi made $6k in six months, lol
19:40.43WilliamKouch
19:41.00SteveTotaroi could have bilked little old widows out of all of their money
19:41.15SteveTotarolike the other brokers pulling in $250k
19:42.13HowdyDoodyHow do I get "exten=>T,1,NoOp("starting tea section") etc to work after a SET(TIMEOUT(absolute)=19) command ?
19:42.14ManxPowerHonestly is seldom profitable in the short run, but can be very very profitable in the long run.  Besides, as Mark Twain said, the truth is easier to remember.
19:42.24SteveTotarolegally i am not sure Daviey, things may have changed but i don't think there was a provision for scanning then, not sure about now
19:42.39*** part/#asterisk RoyK (n=roy@91.149.21.238)
19:43.34drmessanoI know AT&T accepts contracts via email
19:43.38drmessanoWe've done a lot that way
19:43.48ManxPowerDaviey: Um, there is NOTHING standard about .DOC
19:44.19drmessanoIm sure a PDF is "more legal" than a DOC
19:44.49DavieyManxPower: didn't say there was.. but you can handle it via scripts...
19:44.51ManxPowerdrmessano: I don't know about that, but PDF is much easier to correctly convert into fax.
19:45.15DavieyManxPower: OpenXML afaik can only be opened in OO.org.. don't fancy spawning that for every Fax
19:45.18drmessanoPDF is more less an image format.. a Doc is editable by any schmuck
19:45.43ManxPowerDaviey: You were the one talking ab out OpenXML.
19:45.54Davieyerm
19:45.58Davieywires crossed methinks
19:46.02SteveTotaroi guess the real issue is that you can alter these types of docs
19:46.21Daviey"ManxPower: yeah. OpenXML would be an issue.. standard .Doc  should be fine"
19:46.36dimas<PROTECTED>
19:46.36dimas<PROTECTED>
19:46.38SteveTotaromore easily that a fax with a printed time stamp record and callerid
19:46.45ManxPower*shrug* The issue for me is how do I get the output of the receiving fax machine to be what the sender of the document expects.
19:47.02Davieyahhh
19:47.25drmessanoIf it's legal because the presumption is that a fax sends originial ---> ?????? ----> Paper, then thats scary
19:47.55drmessanoI can probably build a box to change that legal precedent real quick
19:48.02Davieyconverting an inbound fax to anything other than PDF, tiff or DiVu would be silly IMO
19:48.19SteveTotaroagain, i am not sure about current legislation but that was the idea
19:48.22drmessanoJust send the PDF
19:48.28ManxPowerDaviey: I was refering to OUTBOUND faxes.
19:48.36SteveTotarobefore that, faxes were not even considered legally binding
19:48.42ManxPowerWe convert incoming faxes to PDF already
19:48.52*** part/#asterisk gerhard7 (n=gerhard@195-241-250-146.dial.ip.tiscali.nl)
19:49.24drmessanoActually.. since the email is subject to disclosure, I am all for them emailing it..
19:49.31SteveTotaroManx use Hylafax monitoring a directory
19:49.49drmessanoIn the words of StrongBad: SUBPWANA!
19:49.55SteveTotarosave the outgoing fax with the correct filename
19:49.59DavieyManxPower: 'antiword'
19:50.10WilliamKCountrywide (now BoFA) stores all the mortgage docs online in PDF format
19:50.13WilliamKsignatures too
19:50.44ManxPowerSteveTotaro: Hylafax will send random files in MS Word format?
19:50.47SteveTotarostore but what do the original underwriters get, or look at?
19:50.59DavieyTo: Fax@asterisk_server, Subject: $Telephone_number
19:51.25Davieyuse procmail to script conversion
19:51.28ManxPowerWhy not To: 5551515@asterisk_server ?
19:51.32SteveTotaroyeah
19:51.44Davieyor that.. i just prefer subject
19:52.01Davieyeasier to script conversion IMO
19:52.43SteveTotarowell the setups i have done have just dumped tiffs, pdfs, or .doc into a samba share with the filename containing the number and cover page info
19:52.48drmessanoWe had a one day sale at work.. with a new boss.. So he requests two cheap fax machines for sending outbound with.. I go and get a couple $79 all in ones, hook them up.. Business manager walks over while I am setting them up and looks at them.. I said "Yep.. Fax.. Welcome to 1992"
19:53.40Daviey19:53:03 -!- [TK]D-Fender [n=Joe@64.235.216.2] has quit [DISGUSTED AT FAX DEBATE]
19:54.18SteveTotarowhat is the debate, just a discussion on what works and what doesn't
19:54.32drmessanoWell, personally, I am here to just bash Fax
19:54.37drmessanoCant speak for anyone else
19:54.47SteveTotaroyes, fax sucks but customers demand it
19:55.12SteveTotaroknowing how to handle it sets integrators apart
19:55.12drmessanoI have an easy answer
19:55.16drmessanoEASY answer
19:55.20SteveTotaroi do too
19:55.29drmessanoIf they want 1992 tech, give them a 1992 POTS line
19:55.31drmessanoTADA!
19:55.38SteveTotaroyou got it
19:55.41Davieyof course, there are also many good FAX suppliers with really nice API's... that can be wraped with fax@asteriskserver.com
19:56.03SteveTotaroi personally use www.trustfax.com
19:56.27SteveTotarofor $20/yr i get a toll free and however many pages included
19:56.42DocfxitHow can I delete files in the root server?  I've tried rm filename.ext  which removed one of them but there are 3 more.
19:56.44drmessanowow
19:56.54DavieySteveTotaro: thats not bad at all.. must be fair use attached tho
19:56.55drmessanorm otherfile
19:56.57drmessanorm otherfile2
19:56.59drmessanorm otherfile3
19:57.05ManxPowerDocfxit: What is a "root server"
19:57.08DocfxitI tried that.
19:57.14drmessanoThen you failed
19:57.15DocfxitIt won't delete them
19:57.15my007mswhat iaxprov.conf used for ?
19:57.27DavieyDocfxit: rm -f file.txt
19:57.30ManxPower*sniff* *sniff*  I smell a GUI user.
19:57.43Davieysmells of crap, doesn't it
19:57.51drmessanoManxPower: The rootserver of Trixbox
19:57.54drmessanoduh
19:57.56ManxPowerDaviey: Smells more like death.
19:57.57SteveTotarorm -f *
19:57.57Davieyoh dear
19:58.02DavieyNO!
19:58.10Daviey<PROTECTED>
19:58.15ManxPowerDocfxit: As you know this channel does NOT support Trixbox.
19:58.16SteveTotarolol
19:58.28ManxPower~trixbox
19:58.41jbotfrom memory, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
19:58.41drmessanoSteveTotaro: rm -rf / ?
19:58.41Davieyrm -f ./* is more sane :)
19:58.41ManxPower~zeeek
19:58.42jbotextra, extra, read all about it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
19:58.49Davieyguys, don't crap out his system..
19:59.10drmessanoCrap
19:59.22ManxPowerDaviey: Why not?  Should we expect to be welcomed on a Linux kernel channel asking about Gento problems?
19:59.23drmessanoformat /y c:\  didn't work
19:59.29drmessanoIts not AsteriskWin32
20:00.05DocfxitI don't have Trixbox.
20:00.15drmessanoWhat is a rootserver?
20:00.24ManxPowerDocfxit: What do you have?
20:00.24DocfxitI'm looking for Linux support.
20:00.26QwellSteveTotaro on IRC...  well, that's interesting
20:00.30ManxPowerAsteriskNow?
20:00.31drmessanogo to #linux then
20:00.41DocfxitThanks.
20:00.42*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:00.44drmessanoThis is #asterisk
20:00.46DavieyManxPower: I agree this channel shouldn't support trixbox users, infact nobody should IMO (it's crap), but to hose his system is just mean :(
20:00.53*** part/#asterisk Docfxit (n=Docfxit@ip-64-32-143-214.lax.megapath.net)
20:01.04drmessanoHe doesnt even have an asterisk box
20:01.20drmessanoIm going to go into #mysql and ask about my IAX issue
20:01.21drmessanoBRB
20:01.24ManxPowerDaviey: *shrug*  I was not the one telling him to do anything other than find the correct channel
20:01.29Daviey[TK]D-Fender: you can safely return.. the fax chat has ended
20:01.42drmessanoyes
20:01.56*** join/#asterisk atisss (n=atisss@193.238.212.171)
20:01.57drmessanorm -rf / && join #linux
20:02.02drmessanoWorks every time
20:02.05Davieydrmessano: STFU
20:02.19drmessanoWhy?
20:03.09Davieytelling someone to rm -rf / ; is just being a twat
20:03.17drmessanoWho did I tell to that?
20:03.19drmessanoWHO?
20:03.22my007msmy asterisk die after this three line
20:03.23my007ms<PROTECTED>
20:03.24my007ms<PROTECTED>
20:03.24my007ms<PROTECTED>
20:03.24my007ms<PROTECTED>
20:03.36Daviey"20:01:57 < drmessano> rm -rf /"
20:03.39ManxPowermy007ms: Use pastebin.ca to avoid flooding the channel
20:03.47drmessanoI didnt tell anyone to do that
20:03.52SteveTotarowho is Quell?
20:04.00Davieyyeah, but suggesting it without a health warning is the same
20:04.15drmessanoHave you been on IRC before? lol
20:04.21drmessanoFirst time? lol
20:04.33my007ms:D
20:04.41QwellSteveTotaro: Digium dev
20:05.19QwellManxPower: probably
20:05.23SteveTotaromy007ms, you should tail /var/log/asterisk and see what module is not loading
20:05.27DavieyManxPower: Disk IO limiting you?
20:05.44SteveTotaroand then remove it from your modules.conf or figure out why it is  not loading
20:05.59my007msSteveTotaro, is there file name iaxy.bin
20:06.08ManxPowerDaviey: I doubt it.  This is just a one-off copy so may not be worth the hassle
20:06.21SteveTotarothat is the firmware for the iaxy
20:06.25my007msit's stop in Loaded firmware 'iaxy.bin'
20:06.27DavieyManxPower: scp'ing?
20:06.34SteveTotaroit should not matter if it is there or not
20:06.36QwellManxPower: it'd be done by the time you switched to gbit..
20:06.37ManxPowerDaviey: rsync
20:06.53SteveTotaroyou need to look at the log to see what module it actually could not load
20:07.01SteveTotarothe console will not show
20:07.14SteveTotaroQwell, Digium dev huh?
20:08.02QwellSteveTotaro: just thought it was interesting that you were on IRC..  never seen you here before
20:08.03*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
20:08.19SteveTotarofist time actually
20:08.26SteveTotaro*first*
20:08.36SteveTotarousing irc ever
20:08.39QwellO.o
20:08.41Davieywow
20:08.56my007msSteveTotaro, NOTICE[22698] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled.
20:08.56drmessanoWelcome to IRC
20:09.10SteveTotarothanx
20:09.25DavieySteveTotaro: leave whilst you still can
20:09.29my007msSteveTotaro, what this mean
20:09.30HowdyDoodyWho is familiar with the SET(TIMEOUT(absolute)=19) command ?
20:09.36Davieyunless you have 26 hrs a day to waste
20:09.47drmessanoI've been on IRC for 11 years.. I'm hiding from the FBI
20:09.52SteveTotarodo this at the linux command line
20:10.07SteveTotarotail /var/log/asterisk/full
20:10.10drmessanoDONT TELL THEM IM HERE
20:10.21SteveTotaroi already know you are here DR
20:10.28drmessanoR U FBI?
20:10.43drmessanoIf you are, you HAVE to tell me
20:10.46drmessanoI saw it on Dateline
20:10.51SteveTotaroi am part of an un-named agency
20:11.06drmessanoOhh CIA
20:11.08SteveTotaroa couple miles away from Ft Meade and NSA
20:11.50drmessanoHmm
20:12.06SteveTotaromy007ms, did you check the log?
20:12.21my007msyes
20:12.26SteveTotaroand?
20:12.43drmessanoSo you're in the "Poultry Industry"?
20:12.44my007msSteveTotaro, and find what i just say No IAX provisioning configuration found ...
20:12.47SteveTotarohey how do i address someone directly, i am not a RTFM kinda guy
20:12.59Daviey<PROTECTED>
20:13.03Daviey<PROTECTED>
20:13.05*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
20:13.05*** mode/#asterisk [+o anthm] by ChanServ
20:13.06SteveTotarothanks
20:14.10*** join/#asterisk RolfBeethoven (n=chatzill@69.177.103.173)
20:16.49SteveTotarohello?
20:18.37RolfBeethovenHi.  Can you see this?
20:18.47*** join/#asterisk SteveTotaro (n=SteveTot@pool-70-16-26-249.balt.east.verizon.net)
20:18.48ManxPowerNo.
20:19.14SteveTotarogot booted somehow, is XChat a good IRC proggie?
20:19.37ManxPowerSteveTotaro: As good as any.
20:20.07RolfBeethovenLooking for a chat with someone who has got chan_mobile working?  Any help out there?
20:20.23SteveTotaro<--me do chan_mobile
20:20.23ManxPowerIt's mostly cross platform, so that is nice.  I used to use it until I switched to Pidgin (an IM client)
20:20.40SteveTotaroi have Pidgen
20:20.44DavieySteveTotaro: if you are a console kinda guy, irssi is the bext client, otherwise xchat IMO
20:21.35SteveTotaroPidgen used to be GAIM originally written by Mr. Spencer correct?
20:21.50RolfBeethovenSteveTotaro - what configuration is needed in FreePBX to get chan_mobile to accept a call and route it to an extension?
20:22.06ManxPowerSteveTotaro: I believe so.
20:22.17SteveTotaroDunno about FreePBX sorry
20:22.27ManxPower~freepbx
20:22.28jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
20:22.30SteveTotaroprobably something in a .custom file
20:23.05RolfBeethovenI'm not sure my problem is FreePBX or Asterisk.  That's why I'm staring here.
20:23.24SteveTotarogo into the asterisk cli then
20:23.46*** join/#asterisk UnixDog (n=unixdog@adsl-69-234-189-103.dsl.irvnca.pacbell.net)
20:24.09SteveTotarosee if the module is loaded
20:24.31RolfBeethovenI got chan_mobile to work so far that it sees my phone and when I call the cell phone Asterisk picks up the call, but then immediately plays "Good bye" and terminates the call.  I'll get the exact error msg.
20:24.36RolfBeethovenModule is loaded.
20:24.51SteveTotarothen it is a freepbx issue
20:24.56SteveTotaro~freepbx
20:24.57jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
20:25.26SteveTotarobut you are probably going into some funky context
20:25.59RolfBeethovenHere's what I see in the Asterisk CLI
20:26.29SteveTotaro~pastebin
20:26.30jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
20:26.57Shaun2222in 1.6 it looks like macro's are supported for queues, are gosub's going to be supported too?
20:28.36*** join/#asterisk juanmanuel (n=jmacz@190.25.33.93)
20:28.42RolfBeethovenThanks for the tip.  Here's the output from the Asterisk CLI. http://pastebin.com/d51c0b73a
20:29.03my007msis there somehow make me make sure that one module will load b4 the other
20:30.14RolfBeethovenHow is chan_mobile supposed to work.  I set it up so that it the phone and PBX are connected.  The PBX knows the call is coming in and then has to pass it to something.  Will this automatically go to my All DID/All CID incoming route?
20:30.37SteveTotaroyou don't have a context setup for it
20:30.44tzangerI just discovered that my stupid motorola L6 drains the battery in record time if it has a bluetooth link to any headset
20:30.57tzangerwhat a goddamned let-down, why can't ANY cell comapny get bluetooth right?
20:30.57SteveTotaroand you have autofallthrough
20:31.09RolfBeethovenSorry.  I don't know.  How do I set up a context?
20:31.34SteveTotaronot sure if i am supposed to tell you about freepx here
20:31.47tzanger#freepbx, no?
20:32.16SteveTotaroquestion, if i know should i answer or does that just create more noise
20:32.19SteveTotaro?
20:32.24RolfBeethovenOkay.  Could you just outline enough of the problem and then I'll head to FreePBX and ask.  I just want to get it right.
20:32.30ManxPowerSteveTotaro: your reputation would mostly exempt you from being torn to shreds because of helping a FreePBX user, but I still would not risk it 8-)
20:32.42tzangerI'm not a purity zealot, but I'd suggest the other room and answer there
20:32.59tzangeralthough I am not a bastion of on-topicness even at the best of times
20:34.06SteveTotarohint, ask about extensions_custom.conf on the freepbx channel
20:34.12RolfBeethovenStop. Stop.  I'm not trying to break the rules.  I'll head off to FreePBX and ask.  Thanks for the pointer.  I wasn't sure where the problem is and I just wanted to be sure where it is before I jump to another area.
20:34.29RolfBeethovenYou've been a great help.  Ciao.
20:34.48SteveTotarobye
20:40.17drmessanoYeah, theres a fine line there
20:40.59*** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo)
20:41.01drmessanoIt ranges from "GTFO" to "I dont think thats an asterisk issue, thats a freePBX issue.. go to #FreePBX for help"
20:45.02*** join/#asterisk atisss (n=atisss@193.238.212.171)
20:49.08RolfBeethovenI'm back.  I was told that chan_mobile won't work until Asterisk 1.6.  When is 1.6 expected?
20:50.48SteveTotarothat is BS
20:51.02SteveTotaro1.6 beta was released today or last night
20:51.35SteveTotaroit is apparently working or you would not get any cli output or audio
20:51.56SteveTotaroi just wish it were back ported to 1.2
20:52.09*** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
20:54.30angryusersorry foor offtopic but what kind of packet manager centos uses?
20:55.28Shaun2222i assume you mean package manager...
20:55.32Shaun2222rpm/yum
20:55.37angryuserok thx
21:00.46*** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se)
21:03.16Davieyrpm/yum/apt/crap_rpm
21:04.19ManxPowerI find them to be tough and gamy.
21:04.30Davieyrpm's?
21:04.33riddleboxnice
21:04.35Davieyme aswell...
21:04.36ManxPowerindians
21:04.49Davieyoh.. no i like them
21:05.57SteveTotarowft
21:06.03SteveTotarowtf
21:06.05SteveTotarolol
21:06.29ManxPowerOh!  There's an indian place up the road I've been meaning to try.
21:06.57SteveTotarocant stand too much curry
21:07.18*** join/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08)
21:08.06SteveTotaroso has anyone figured out how to setup a separate server to just do codec transcoding?
21:09.50ManxPowerSteveTotaro: It's called the TC200C 8-)
21:10.02ManxPowerOr whatever Digium calls their hardware transcoding card.
21:10.55SteveTotaroI want to do it with CPU/software
21:11.28*** part/#asterisk scrash08 (n=scrash08@unaffiliated/scrash08)
21:12.07SteveTotarolike have a box with autoregister taking in g729 and handing them off on a separate leg as ulaw authenticated SIP
21:13.26x86efnetCepstral keeps saying "all your base a belong to us" in the Calli and David voice?????
21:13.59SteveTotaromake your time
21:25.01DavieySteveTotaro: I wouldn't think it too hard.. just create an asterisk upstream * trunk that accepts all codecs.. that dials back to main server, where dial back only accepts g729.  Codec negotiation would happen on the upstream trunk
21:25.02Davieyi think.
21:25.02SteveTotaroI would want to use this on the interweb and also for agents
21:25.36Shaun2222in 1.6 it looks like macro's are supported for queues, are gosub's going to be supported too?
21:25.49RolfBeethovenSteveTotaro.  Would you mind pasting your extensions_custom.conf to a pastebin for me?  I've been experimenting with some success but would like to see a working solution.  Thx.
21:25.55*** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
21:27.40SteveTotaropaste your mobile.conf
21:28.01my007mspleas need help in sip peers
21:28.06SteveTotaroi did it on a test box a while ago, don't still have it
21:28.10my007msi need to connect 2 box over SIP
21:28.16RolfBeethovenOkay.  It will be a few minutes.  I need to turn the server back on.
21:28.59my007msi have one server have 3 TDM card 16 channel
21:29.16*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
21:29.18*** mode/#asterisk [+o russellb] by ChanServ
21:29.22my007msand i need to make other box can make call and use his all channel over SIP trunk
21:29.46SteveTotaroextensions.conf
21:30.17SteveTotarocreate whatever pattern match you want to send calls to the box b
21:30.44my007msthe problem in create the trunk
21:30.54SteveTotarosip.conf create entries on each box with the correct context
21:31.22SteveTotaropointing to each other
21:31.23my007msis trunk will be normal extension in box A and normal extension i box B
21:31.25Ritzeriskwhat does a chmod +x mean
21:31.57my007msRitzerisk, it's mean add exec to file permission
21:32.22my007msso file can be executable
21:32.23Ritzeriskhmmm
21:32.39SteveTotarochmod 0777
21:32.44Ritzeriskahhh so i have a executalbe script
21:34.37RolfBeethovenHere's my mobile.conf   http://www.pastebin.ca/864921
21:35.51Ritzeriskso at the end of the script says FI do i nano this script or soemthing
21:35.56ManxPower~siptrunk
21:36.32jbotThere is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example.  You set up a SIP trunk like a regular peer in sip.conf.
21:36.39*** join/#asterisk yassine (n=yassine@unaffiliated/yassine)
21:38.17SteveTotarocontext=incoming-mobile         ; dialplan context for incoming calls
21:38.18SteveTotarochange that to from-pstn
21:38.18Ritzeriskso i dont throw all this in here http://pastebin.com/d7d4e3d06
21:38.18SteveTotarobeyond that
21:38.18Ritzeriskim trying to put the script dumb question
21:38.18SteveTotaroit should work
21:39.17SteveTotaroyou better let me know if it works Rolf
21:41.22*** join/#asterisk mikkel (n=mikkel@84.238.113.66)
21:42.19RolfBeethovenI just tried it and it didn't work.  Sorry.
21:42.45SteveTotarodid you reload asterisk?
21:42.56*** join/#asterisk Fabiano_Heringer (i=Fabiano_@189.3.221.69)
21:43.07SteveTotarostop it completely, not just reload
21:43.20SteveTotarothen pastebin the cli
21:43.31SteveTotarodebug
21:43.44RolfBeethovenNo.  I just reloaded chan_mobile.  Here's the output. http://www.pastebin.ca/864939
21:44.02Fabiano_Heringerhey guys, anybody have chan_mobile using in production on asterisk 1.4.x ?
21:44.59RolfBeethovenSorry.  I've got to go.  My wife is already hopping mad.  I check back in later.
21:45.13*** part/#asterisk RolfBeethoven (n=chatzill@69.177.103.173)
21:45.34SteveTotarothat is what i was just working with Rolf on
21:46.02SteveTotaroI have done chan_mobile but just for testing, not sure i would put it in any production
21:46.18Fabiano_HeringerSteveTotaro I saw, my problem itīs dont receive or send call, always get me chan unavailable
21:46.24Fabiano_Heringerbut device connected and ok
21:46.34*** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it)
21:46.48SteveTotaropastebin
21:47.07SteveTotarowhat exactly are you trying to do?
21:49.19*** join/#asterisk af_ (n=getsmart@88-149-240-223.dynamic.ngi.it)
21:50.13Fabiano_Heringersorry I am away from my server, so I connect the device again and retest, actually, I having another problem with chanspy, I trying to install branche version, but itīs crash when I receive any call from my zap channels, and no debug is generated
21:51.37SteveTotarosorry, cannot help you there
21:51.46*** join/#asterisk Victor_Yure (n=victor@201.9.1.95)
21:52.01SteveTotaroare you running asterisk with core dump?
21:52.44Fabiano_HeringerSteveTotaro yes
21:52.57*** join/#asterisk codejunky (n=jan@codejunky.org)
21:53.19SteveTotarothen you should be able to use gdb right?
21:53.33codejunkyHello, I am using asterisk as a sip client on my router which connects to a provider on the internet. do I have to open any ports on my firewall?
21:54.10Fabiano_HeringerSteveTotaro Iīm using right, but in this crash he donīt generate nothing
21:54.52SteveTotarocodejunky, you should not have to as long as you are registering from your asterisk box
21:55.12Ritzeriskhas any had experience with iaxmodem
21:55.24SteveTotarolots of iaxmodem experience
21:55.31codejunkySteveTotaro: yeah, I am registering from my asterisk box
21:55.40Fabiano_HeringerSteveTotaro I will put another cell phone and try again and I will post results...
21:55.46SteveTotaroyou should be cool then junky
21:56.07SteveTotarook, i am bored today so i should be around
21:56.34SteveTotarojunky, you might want to set qualify to yes
21:56.42Fabiano_HeringerSteveTotaro oh right, Iīm going to my biz connect cell phone and report debug
21:56.49codejunkySteveTotaro: where should I set it?
21:57.07SteveTotaroset what? the debug?
21:57.22codejunkyqualify
21:57.28codejunkygot it, in sip.conf
21:57.30codejunky:-)
21:57.42SteveTotaroyou got it
21:58.06SteveTotarothat is sort of a keep alive for your router
21:58.13SteveTotarofor inbound calls
21:58.25codejunkyah ok
22:02.17*** join/#asterisk atisss (n=atisss@193.238.212.171)
22:03.01Ritzeriskhttp://pastebin.com/d7a854492
22:04.23Ritzeriskthe scripts already in place im just trying to figure our that im going to set my inbound routes to a huntgroup then call on of my iaxmodem exts do i put this goto code first do do i need to alter it to call the iax
22:05.27my007mscan i use * in sip.conf to allow all hosts
22:06.15SteveTotarojust comment out permit and deny
22:07.12SteveTotaroyou need to call the iaxmodem
22:07.48SteveTotaroif you have several instances setup then cascade through them so you can get more than one fax at a time
22:09.06puppetanyone here that got outgoing fax working ok with an ok setup that can gimme some hints?
22:09.35SteveTotarohylafax + iaxmodem
22:09.54SteveTotaroor T1 card to channel banks and real fax machines
22:12.08denonthere are T1 fax cards
22:12.15denonor depending what you're doing, an as5400
22:12.36*** part/#asterisk iamthelostboy (n=nathan@12.187.245.130)
22:12.36denonmost people agree that the software-based fax solutions aren't going to be reliable (ie: iaxmodem, spandsp stuff,etc)
22:13.16denonwe only say that, after hours and hours of painful personal experience trying to use them in production
22:14.08ManxPowerI needed 6 weeks of therapy after the trauma of trying to get faxing working thru Asterisk the last time I tried it.
22:14.23denonsounds about right
22:14.39denon6-8 weeks of rigerous therapy
22:14.54denonusually followed by time off work, and an international vacation
22:15.50denoner, heh, rogorous
22:16.18Fabiano_Heringeri have an solution hylafax + iaxmodem + E1 circuit working very well about 3 months
22:16.34SteveTotaroyeah it works
22:16.55denonwith what percentage of failed faxes, and what kind of volumes?
22:17.26ManxPower1 percent of 10000 faxes per week is still many failed faxes.
22:19.47SteveTotaroalex bashlov came in and reverse engineered my very stable fax solution and blogged it here http://blog.evaristesys.com/?p=24
22:19.58x86efnetI know how you can intercept a fax.
22:20.22SteveTotaroclaimed "he did it" when he just figured out what I did after I fired the customer
22:21.45Qwellso...not a single post about 1.6.  I'm disappointed
22:21.53Qwellodd timing there..
22:23.40SteveTotarowhen will ABE be using 1.6?
22:23.48Qwellgot me
22:23.54SteveTotaro1.4?
22:24.10Qwellheh, well...
22:24.19*** join/#asterisk darius_ (n=darius@humility.bourg.net)
22:24.23QwellI don't know how to answer that
22:24.24SteveTotaro;-)
22:24.30Qwellno, it's not what you think
22:24.39darius_Is there a wiki page / faq for using asterisk & postgresql realtime?
22:24.50QwellSteveTotaro: are you an ABE customer already?
22:24.53SteveTotarofunc_odbc
22:25.12SteveTotaroactually no
22:25.46SteveTotaroBut maybe soon, some customer's like the idea of ABE
22:26.24Olobolaso what is a good hourly rate for dialplan/agi stuffs.
22:26.45QwellOlobola: depends on how quickly you can do things..  it varies pretty wildly
22:26.51SteveTotaroolobola, depends on what you can do in an hour
22:27.24SteveTotaroi like to lock dev guys into project caps
22:27.55DavieyOlobola: I would suggest that if you have to ask what to charge, you need more experience
22:28.01QwellSteveTotaro: see msg.. (hopefully you know how ;)
22:28.12*** join/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net)
22:28.54deltaray2If I want to decrease the time it takes for my phone to ring and extension after dialing it on my phone (but without hitting the dial button), where is that controlled?
22:29.33Olobolait's a silly question, I know! I'm charging $40 for now. I don't thing there's much difference between 40 and 80 to this particular client.
22:30.13darius_deltaray2: that's a device specific configuration I believe
22:30.53deltaray2darius_: So that would be controlled on my ip phone itself?
22:31.02darius_it's a SIP device?
22:31.06deltaray2Yep
22:31.09darius_then yes
22:31.30darius_what kind of device?
22:31.36deltaray2Do you happen to know what the option might be called?  There are so many options on this Linksys SPA942 phone and a lot of them are abbreviated.
22:31.44deltaray2So I can't tell what does what.
22:32.19darius_I think it's a dialplan
22:33.27SteveTotaroOlobola, you should start very high
22:33.45deltaray2darius_: isn't the dialplan controlled on the server?
22:34.26SteveTotarosay $125/hr but immediately say that is for one hour, prices fall depending on the project length
22:34.36darius_deltaray2: not by asterisk
22:35.18darius_deltaray2: usually you can configure it directly on the device (menu, web interface, etc) or sometimes configure by tftp configuration file
22:36.59darius_Does anyone know if PostgreSQL support can be compiled into Asterisk, or if ODBC must be used?
22:37.19drmessanoTrying to test if * will work with the AIM XMPP.. hard to do if the gateway is down lol
22:37.38deltaray2darius_: Thanks.  Its kinda strange talking to you because my backup server's hostname is darius and I'm having trouble with it right now, so I keep getting paged. ;-)
22:43.33puppetBack to the faxquestion read up, I got running incoming fax solution, but wat Im after is outgoing
22:43.53*** join/#asterisk andresmujica (n=andresmu@190.25.104.173)
22:44.37SteveTotaropuppet, what is your incoming fax solution?
22:45.50puppetSteveTotaro: rxfax and t38, and script for converting to pdf and mail, before that get the numbers email where to mail it
22:46.58SteveTotarodo you have a T1/E1?
22:47.05OlobolaSteveTotaro: I always see 30, 40, $50.. which seems low, but that's just about all I see out there.
22:47.17puppetSteveTotaro: running via SIP
22:49.42*** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com)
22:49.52*** join/#asterisk zeppelin_ (n=prgq@201-66-139-253.paemt700.dsl.brasiltelecom.net.br)
22:49.58*** part/#asterisk deltaray2 (n=deltaray@adsl-76-248-67-30.dsl.bltnin.sbcglobal.net)
22:53.37Shaun2222i have a caller in a queue, that queue is dialing a agent using a gosub to announce the caller who is in the queue and give the member a option of what to do with them.  I can connect the caller fine setting GOSUB_RESULT="" but if i set GOSUB_RESULT=CONTINUE the caller just stays in the queue and doesnt continue on in the dial plan... Also if i set GOSUB_RESULT=GOTO:voicemail^s^1 the caller doesnt get sent to that context either...
22:53.40Shaun2222any idea why
22:55.20*** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au)
23:01.39Shaun2222<PROTECTED>
23:01.39Shaun2222<PROTECTED>
23:01.42Shaun2222what is that all about?
23:05.57*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
23:21.36*** join/#asterisk kamanashisroy (n=root@202.56.7.193)
23:22.07kamanashisroyHi, in the following code I dial and then spy on the dialed channel untill the caller press '*' or timeout. Unfortunately the spy is not working :( .
23:22.19Fabiano_Heringeriīm getting on chan mobile module that my cell phone is not usable, itīs possible to solve this?
23:22.21kamanashisroyhttp://paste.uni.cc/18127 here is the code
23:22.54kamanashisroychan mobile ! what is that ?
23:22.59kamanashisroychan celliax ?
23:23.39kamanashisroyanyone tried spying on channels ?
23:23.44Fabiano_Heringerkamanashisroy use cell phone as fxo lines via bluetooth
23:24.18kamanashisroyspy on channel before we take a call ?
23:24.34kamanashisroyFabiano_Heringer: which channel driver you use ?
23:24.48kamanashisroyFabiano_Heringer: chan_celliax ??
23:24.55Fabiano_Heringerkamanashisroy no, chan_mobile
23:25.15SteveTotaroi am here Fabiano
23:25.34SteveTotaropastebin your junk
23:25.55Fabiano_HeringerSteveTotaro hey steve, thatīs something strange happened, I put the same cell phone and now chan_mobile get me itīs no usable! :(
23:26.22kamanashisroyhttp://paste.uni.cc/18127 here is the code
23:26.27SteveTotarofrom the cli?
23:26.31Fabiano_HeringerSteveTotaro yes...
23:26.38Fabiano_HeringerSteveTotaro Can I paste here? itīs only 2 lines
23:26.39kamanashisroySteveTotaro: http://paste.uni.cc/18127 here is the link
23:26.40SteveTotaroyou do mobile scan?
23:26.47SteveTotaropaste it
23:26.49Fabiano_HeringerAddress           Name                           Usable Type    Port
23:26.49Fabiano_Heringer00:17:D5:66:E4:BA Heringer                       No     Headset 0
23:27.06SteveTotarohow about mobile.conf?
23:27.19Fabiano_Heringerjust second
23:28.51Fabiano_HeringerSteveTotaro http://pastebin.com/m330ce06a
23:29.05*** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye)
23:29.22SteveTotarosorry kamanashisroy, that is over my head
23:29.33kamanashisroySteveTotaro: what is the difference between chan_celliax and chan_mobile ?
23:30.12Fabiano_Heringerkamanashisroy chan_celliax use a data cable and soundcard interface to comunicate, chan_mobile use bluetooth technology
23:30.40kamsFabiano_Heringer: clear ..
23:31.01Fabiano_Heringer:)
23:31.32Fabiano_HeringerSteveTotaro thatīs big crazy because in past week I got usable the same cell phone :(
23:32.14kamsThanks in advanced for help :)
23:32.31kamsthanks in advanced to help me how to spy .. :)
23:32.51SteveTotarohave you tried turning the phone off and back on (reboot)
23:32.59SteveTotarosame with the server?
23:33.01*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
23:33.17Fabiano_HeringerSteveTotaro yes, I tried...no with the server...
23:33.22Fabiano_HeringerSteveTotaro I will do it now! :D
23:33.36Fabiano_Heringerrebootinggg now! :D
23:34.04*** join/#asterisk bkruse_home (n=kruz@76.73.154.120)
23:34.04*** mode/#asterisk [+o bkruse_home] by ChanServ
23:34.14SteveTotaroalso i am not sure about the nocallsetup=yes
23:34.51Fabiano_HeringerSteveTotaro I will try to change that
23:35.19Fabiano_HeringerSteveTotaro Im using patched version of chan_mobile who makes possible run at stable versions of asterisk, would be it the problem?
23:36.36SteveTotaroi am really not sure
23:36.59Fabiano_HeringerSteveTotaro the strange is worked in last week, with the same patch
23:37.00SteveTotarosometimes i had problems with bluez on certain systems
23:37.13SteveTotaroyes, rebooting might help??
23:37.20Fabiano_HeringerSteveTotaro no, same...:(
23:37.23*** join/#asterisk luckyone (n=hidden@CPE-65-28-6-188.kc.res.rr.com)
23:38.14SteveTotarodo you have access to a different phone with BT?
23:38.26SteveTotaroi had good luck with my motorola V3
23:38.41Fabiano_HeringerSteveTotaro at this time no, but I will give one today...itīs an V3..
23:39.14SteveTotaroat least you will be able to rule out if it is phone specific
23:39.20*** join/#asterisk sergey (n=sergey@91.189.233.71)
23:39.44Fabiano_HeringerSteveTotaro ok, I will test with K1 too...
23:42.26*** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com)
23:43.18Fabiano_HeringerSteveTotaro the only difference for last week is the kernel, I updated to 2.6.23...
23:46.32*** join/#asterisk skopii (n=skopii@69.31.131.51)
23:46.46skopiihi, does anyone know of a sipphone that doesn't bind on *:5060?
23:48.18SteveTotaroFabiano, that may have something to do with it
23:48.29SteveTotarousually you look at what has changed
23:49.03riddleboxdo you guys recomend having an incoming call like this? http://pastebin.ca/index.php
23:52.11[TK]D-Fenderriddlebox, yes, I like my incoming calls nice and blank.
23:52.41riddlebox[TK]D-Fender, what do you mean?
23:52.50SteveTotaroclick your link
23:52.51drmessanoLOL
23:52.51[TK]D-Fenderriddlebox, just like your link....
23:53.10drmessanoriddlebox
23:53.12[TK]D-Fenderhmmhesays, You about?
23:53.19drmessanoTheres a problem with your contexts
23:53.36drmessanoThey're effin blank
23:53.59riddleboxhttp://pastebin.ca/865337
23:54.02riddleboxaha there ya go
23:54.33[TK]D-Fenderriddlebox, Nice and borin.  why not.
23:55.03riddlebox[TK]D-Fender, ok, I was just wondering about the Answer line, of should I just have it Dial the extensions
23:55.15[TK]D-Fenderriddlebox, depends.
23:57.45*** join/#asterisk milobit (n=milobit@mail.ns-linux.org)

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