irclog2html for #asterisk on 20051125

00:00.17manywell, GET DATA <file> [timeout] [maxdigits]
00:00.24manyanswer your question yourself.
00:00.40MindSparkbut no matter what I do it only sends back one digit
00:01.17manythen you probably encountered timeout
00:01.30*** join/#asterisk Nix (n=Nix@81.213.125.220)
00:01.39MindSparkhmm, I'll try to set that to 50 or so
00:02.16*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
00:07.31*** join/#asterisk MrbBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net)
00:08.17*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
00:09.25*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
00:14.03Dr-Linuxhow many lines/login can be used in cisco 7960 ip phone?
00:16.18nahireanexten => s,2,Background(/var/lib/asterisk/mohmp3/bf.mp3) should be a valid extension and function right?  Asterisk complains however that it cannot find this file..
00:16.27Dr-Linuxcan we telnet the Cisco 7960 phone ?
00:20.18*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
00:25.41tzafrir_laptopnahirean, does asterisk support mp3? try without the extension.
00:26.21nahireantzanger: is there a quick CLI command I can do to check?
00:26.29nahireansilly nick complete
00:26.43nahireantzafrir_laptop: the on hold music is mp3 format, but I cant get that stuff to play either
00:39.41*** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net)
00:42.40tzafrir_laptopnahirean, do you have the module format_mp3 ?
00:42.49tzafrir_laptopit is part of asterisk-addons
00:44.24shmaltzis this ther right way to set cidname in 1.2:
00:44.26shmaltzSet(CALLERID(NAME))=${ARG7})
00:44.41shmaltz?
00:54.55*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
00:57.28*** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net)
00:58.02nahireancls
00:58.05nahireanmistell
00:58.20*** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com)
01:06.49*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
01:09.35*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
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01:10.29asterboyPolycom wants you to take a test before you sell...oh please!
01:10.58asterboyWhat a joke...guess I'm stuck buying from eBat.
01:11.54_DAWdoes anyone here use the page cmd.  I am trying to determine if I can use alert_info with it..
01:12.19*** join/#asterisk JacquesL (n=jl@ool-44c1650d.dyn.optonline.net)
01:12.33asterboyAnyone know of a vendor who realises that most people who would even consider buying an IP phone is a geek?
01:14.01*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
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01:18.48shmaltzsdyrtboy, can you explain this question?
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01:19.23*** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
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01:25.56*** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org)
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01:27.54AmbroseAnyone had success with fax2email? I'm trying to figure out how to get Asterisk to accept faxes, then e-mail them
01:39.08*** join/#asterisk alphadad (n=derd@24.83.96.214)
01:43.12alephcomAmbrose:  I would use hylafax for that
01:44.42*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:44.54*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:52.28h3xrxfax dosent work for shit
01:53.57mog_homei have a new fax thing
01:54.02mog_homeits called E-Mail
01:54.20AmbroseOh I was just following the spandsp/fax howto on asteriskguru's site
01:54.30h3xdude
01:54.35h3xspandsp fax receive dosent work
01:54.46h3xmost of the time
01:54.50{zombie}I just love the animosity towards fax users in this channel
01:54.52AmbroseGeez.
01:54.56{zombie}spandsp fax receive works fine
01:55.31mog_homelol zombie
01:55.37mog_homewell its just not something asterisk does well
01:55.42AmbroseSo this hylafax is supposed to work?
01:55.47mog_homeand its not something the majority of asterisk users want
01:55.50mog_homei mean people want it
01:55.55mog_homebut most would rather have other stuff done
01:56.07mog_homeand things like hylafax exist
01:56.10mog_homethat rock
01:56.44Qwell[]You know what I use for fax machines?
01:56.55Qwell[]rather, what I use fax machines for.  Target practice.
01:58.53h3xthere is an assload of clients for hylafax
01:59.10{zombie}mog_home: the problem is, if you have a digium or sangoma e1/t1 card, how do you use hylafax with it?
01:59.19{zombie}if that's your only line into the building, your options are rather limited
01:59.26AmbroseSo does anyone have a good howto on how to setup hylafax?
01:59.31{zombie}I would love to use hylafax instead
01:59.33mog_homeyeah
01:59.38{zombie}in fact I even tried hylafax via iaxmodem
01:59.42{zombie}because someone suggested that
01:59.45{zombie}and still can't send anything
01:59.52mog_homealso the fax people havent funded any fax development in the community
01:59.55mog_homethey just want it
02:00.01mog_homei mean look at our pri stack
02:00.13mog_homea lot of that work was done by people who needed it and got it done
02:00.21mog_homeor app_dictate and other things etc
02:00.39{zombie}yup
02:00.52{zombie}I'm just saying, telling people to "just use hylafax" really isn't helpful
02:00.55*** join/#asterisk TestMaster (i=Computer@66.244.235.210)
02:00.59mog_homeagreed, but its oss
02:01.08TestMasterhello all question can i have asterisk send a dial tone. to my ata?
02:01.09mog_homeput your money or efforts where your mouth is
02:01.21mog_homeyour ata probably generates its own testmaster
02:01.28mog_homeasterisk can gen dial tone though
02:01.33tzafrir_laptopTestMaster, the ata generates the dialtone, not asterisk
02:01.37TestMastermog_home no it doesn`t i just tried
02:01.47mog_homeits not registerd to asterisk i bet
02:02.02TestMastertzafrir_laptop for some reason the developer of this hardware didn`t set it up to produce a tone
02:02.04mog_homemost atas dont gen dial town unless its regged
02:02.11TestMastermog_home ya it is
02:02.18mog_homewell you could set it to immediate
02:02.21mog_homeand have it disa
02:02.21tzafrir_laptopTestMaster, what ATA is it?
02:02.27mog_homethat would give it dial town
02:02.39Qwell[]dial town?
02:02.42TestMastertzafrir_laptop not sure the make of it
02:02.51TestMastermog_home set what to immediate?
02:02.58tzafrir_laptopHow does it connect to asterisk?
02:03.00TestMastertzafrir_laptop there is no make on it.
02:03.05Qwell[]mog_home: I want to take you to...dial town
02:03.32mog_homei need a typist
02:03.51tzafrir_laptopsomeone who makes typos?
02:03.59TestMastertzafrir_laptop the normal way any sip device. its connected to asterisk fine.
02:04.20mog_homecan you set a time out on the dialing mechanism testmaster?
02:04.24TestMasterits made by Epic Systems
02:04.29mog_homeyou will want to set dial timeout to instant basically
02:04.37TestMastermog_home no it doesn`t give you any options like that
02:04.42mog_homeand then it will send a dial with something and then you could have it go to disa
02:04.46mog_homewell you are probably sol
02:04.57mog_homedont use shit hardware and your life will be better
02:04.58TestMasterOk thanks
02:05.13mog_homesorry
02:05.16tzafrir_laptopTestMaster, if it connects to Asterisk through voip (e.g: sip) then asterisk doesn't know about the analog phone and has no way of figuring it needs to send dialtone
02:05.32mog_homeit can send dialtone once its connected in a call
02:05.33mog_homevia disa
02:05.35TestMasterits good hardware for the most part. all but the dial tone part. i tried it on a dial up connection. and had it connect to asterisk. it sounded perfect
02:05.38mog_homebut it has to dial first
02:05.51tzafrir_laptopTestMaster, maybe the lack of dialtone indicates that the device is "not connected"?
02:05.55mog_homeso you could in theory setup dial plan to just have it go straight to said disa
02:06.04mog_homebut usually tzanfrir is right
02:06.06TestMaster<PROTECTED>
02:06.10TestMasterits connected :-)
02:07.16TestMastermog_home ok thanks... i am going to contact the maker. he should have some input.... and thanks tzafrir_laptop
02:07.40mog_homeman i wish i was an op status file
02:07.47file:)
02:07.53fileI poke people I like
02:07.53mog_homeand i would config my irc client to kick people who poke me
02:08.04tzangerhaha
02:08.17mog_homeor wait not kick
02:08.19mog_homejust mute
02:08.21file:P
02:09.09mog_homeand then i would say
02:09.13mog_homei mute people i like ^_^
02:10.00SwKanyone remember a bug in the past couple of months in chan_sip that causes it to hang or dead lock on a reload chan_sip >
02:10.09SwKfile
02:10.12SwKmog
02:10.49mog_homeyes
02:10.51mog_homei think
02:10.55fileon a reload chan_sip? mmm yeah
02:11.08SwKi'm tickling it
02:11.12SwKbetter find the patch
02:11.19mog_home?
02:11.23fileis it under a heavy amount of peers?
02:11.28SwKyeah
02:11.34SwK~350
02:12.07mog_homethe bug not fix it for you swk
02:12.16SwKi dunno
02:12.19fileis qualify turned on? maybe chan_sip is sending a packet to every one on a reload to see if they're alive
02:12.23SwKi'm looking at mantis right now
02:12.35SwKqualify is on on most of them
02:12.39SwKand it doesnt do ie every time
02:12.43filefunkjy
02:12.44fileer funky
02:12.47SwKonly everyone once in a while
02:12.54mog_homeasterisk -g
02:12.56mog_homeis your friend
02:13.11Qwell[]run -g is moreso
02:13.26SwKyeah well its not segfaulting
02:13.34Qwell[]strace?
02:13.35mog_homethen grab a core
02:13.45mog_homeast_grab_core is your friend too
02:14.00fileSwK: attach with gdb to the PID and see what it's doing when it occurs?
02:14.08SwKi havent tried that
02:14.14SwKi'll do that next time
02:14.22mog_homeast_grab_core
02:14.28mog_homeand then let some one look at it
02:14.31SwKi already restarted as its a busy ass box
02:14.33mog_homealso do an unoptimized build
02:14.39SwKno
02:14.54mog_homewell you probably wont be able to debug the optimized one
02:14.55fileI hate sending off quotes for stuff
02:14.57*** join/#asterisk Cresl1n (n=matt@24.214.255.160)
02:14.59mog_homebut whatever floats your boat
02:14.59SwKnot on a heavy call router w/ 300+ peers
02:15.40SwKmog dont make me drive over to you office on monday and slap you with a vi manual
02:16.01mog_homego right a head swk
02:16.09SwKheh
02:16.13mog_homebut to fix problems you sometimes have to make it worse
02:16.23mog_homebecause i dont share the stable wand
02:16.27SwKcant do that on this box
02:16.45SwKwould bt nice if i could
02:16.48mog_homewell then dont run sip show peers
02:16.59SwKi didnt run sip show peers
02:17.05SwKi said sip reload
02:17.16mog_homeor reload
02:17.17mog_homemy bad
02:17.36SwKand the system required reload or you cant add/delete peers
02:17.41SwKits not running realtime
02:18.34mog_homewell you are destined to keep having problems
02:18.40mog_homegood luck with the core dump
02:18.59SwKatleast until 1.2.x stabilizes and I can port the custimizations over
02:20.03*** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp)
02:20.22Pikorohey, anyone got any idea how i can patch the zaptel drivers to work properly in Japan?
02:20.49mog_homeset to t1 i believe
02:21.07SwKfor a j1 or for POTS?
02:21.13Pikoroi remember seeing something about spandsp or a patch to the zaptel drivers to make them work
02:21.15Pikoropots
02:21.49mog_homegood luck
02:21.50Pikoroit works.. sort of.. sometimes there are hangup detection issues and oddball times where asterisk won't answer the incoming call
02:23.43*** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239)
02:23.49Pikorosup fender :D
02:23.50[TK]D-FenderWhee!
02:24.04[TK]D-FenderMy first time using BitchX from CONSOLE!
02:24.09[TK]D-FenderI feel so... l33t
02:24.17Pikorohaha
02:24.29[TK]D-FenderA necessary evil as I am doing a server upgrade on this box and can't stat X :)
02:24.50[TK]D-FenderAnd my new Sangoma S518 = success!
02:24.59Chujiat least you didn't do it as r00t
02:25.03[TK]D-FenderCorrect!
02:25.15[TK]D-FenderI'm DUMB, not STUPID!
02:25.20*** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com)
02:25.23*** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
02:25.26[TK]D-Fender:O
02:25.37[TK]D-Fenderfile : it works like a charm :)
02:25.47*** join/#asterisk jahani (n=k@adsl-211-43-192-81.adsl.iam.net.ma)
02:25.50Pikorogotta get this japanese support working
02:25.54kimcgreets
02:26.13[TK]D-FenderPikoro : Oh yeah.. now I remember what I helped you with!  How's it going?
02:26.33kimcanyone know how to prepend a 1 to outbound numbers ?
02:26.42jahanihi
02:26.51[TK]D-Fenderkimc  : just shove the "1" in front
02:27.02jahaniwhat type of card i have to buy to make ip pstn switing with asterisk ?
02:27.06PikoroDial(1${EXTEN})
02:27.09Pikoro?
02:27.20Pikorogoing pretty good
02:27.27kimcRight but how about without requiring dialing the 1 every time ?
02:27.37kimcNuFone requires this
02:27.44Pikorojust having some oddball problems with hangup detection... i think i need to use spandsp or something to get callerid working
02:27.52kimc1 + 10 D
02:27.56Pikorosince callerid in japan requires like a 2 second hangup or something really wierd
02:28.06[TK]D-FenderPikoro : you need to refer to the "tech" that is being accessed like "exten => _9x.,1,Dial(ZAP/g1/1{EXTEN:1))
02:28.48[TK]D-Fenderhangup detection is something that's always hard with analog lines, esp Zap.
02:28.49camonzhi
02:29.10[TK]D-FenderNot sure about callerid though
02:29.18camonzi was wondering if there is any distinction between a SIP Server and a SIP Agent like xteen
02:29.25camonzwhen configuring them on sip.conf
02:29.40[TK]D-Fenderkim : copy your dial line here and we'll mod it for you
02:29.52kimccool stby..
02:30.10Dr-Linuxcan we telnet the Cisco 7960 phone ?
02:30.24[TK]D-Fendercamonz : Well you normall have a "register" line in SIP.conf for registering * to another SIP server (like a voip provider)
02:30.39Dr-Linuxand does it have all the network configuration command to load the SIP firware?
02:30.51kimc_61XXXXXXXXXX,2,Dial(IAX2/me@NuFone/${EXTEN:1},20,tr)
02:30.58Dr-Linuxfirmware*
02:31.01*** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net)
02:31.04bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
02:31.06*** join/#asterisk Thazza (n=me@203.80.44.200)
02:31.36[TK]D-Fenderkimc : why the 61 prefix?
02:31.46camonz[TK]D-Fender: what about if i want asterisk to behave like a server,
02:32.16kimcUsing a leading 6 to choose which outbound and Nu
02:32.19camonzit is my first setup,
02:32.29kimcNufone requires the leading 1 digit
02:33.05*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
02:33.10kimcThat is.. send the 1 to Nufone
02:33.44kimcEXTEN:1 <-- eats the leading 6
02:36.51kimcThere are some references to 'prefix'
02:37.28*** join/#asterisk froguz (n=froguz@97-134-222-201.adsl.terra.cl)
02:39.42*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
02:40.29kimcOne way: http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html
02:40.47kimc'Step 5'
02:42.35kimcDunno if 'prefix' is still valid.. this example also uses StripMSD which doesn't work anymore I don't think
02:43.46*** part/#asterisk [hC] (n=hardcore@209.153.195.139)
02:45.13[TK]D-Fender1 sec
02:45.44jahaniis there prepaid calling card system for asterisk ?
02:46.49mog_homeastcc
02:47.10jahanifrom where i can get it?
02:47.41[TK]D-Fenderkim so basically you wan to do 6xxxxxxxxx to dial a 10-digit # and have it add the 1 for you?
02:47.58froguzwhat if i get a TDM400p (w/ 2 fxo), do i have to delete ztdummy?
02:48.31kimcExactly just eat the 6 and add the 1 plus the 10 digits
02:48.57*** join/#asterisk TestMaster (n=workingg@66.244.235.222)
02:49.11mog_homecvs co astcc
02:49.44TestMasterHello all question with asterisk, if the user has a unknown phone number, can i have it so asterisk will ask them for there number, otherwise it will allow them to call through if they have caller id number showing?
02:50.10[TK]D-Fenderthen - exten => _6XXXXXXXXXX,2,Dial(IAX2/me@NuFone/1${EXTEN:1},20,tr)
02:50.22[TK]D-Fenderjust shove the 1 in there
02:50.51kimcAh right.. great
02:51.40jahaniok thank you mog_home
02:52.04jahanihttp://store.digium.com/product_view.php?category=2&product_code=TE406P this card is beter or cisco gateway is good?
02:53.20TestMasteranyone?
02:53.34benjkTestMaster: yes
02:53.42mog_hometestmaster you can
02:54.07mog_homeand jahani get a 406 they rock ^_^ <I work for digium disclaimer>
02:54.50TestMasterbenjk or mog_home could you recommend what to look under... on the wiki?
02:55.24benjkon the CLI type show application PrivacyManager
02:55.31mog_homeyay
02:56.00TestMasterbenjk oh ok thanks
02:56.00kimc[TK]D-Fender you rock :)
02:56.37Kattypaper scissors.
02:58.07benjkseen ~wasim
02:58.15benjkdamn
02:58.21benjk~seen wasim
02:58.25jbotwasim is currently on #asterisk (3d 21h 2m 31s).  Has said a total of 34 messages.  Is idling for 8h 51m 14s
03:01.27*** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net)
03:03.16jahaniany one speak french ?
03:03.32benjkjahani, why?
03:04.44ceph__directed to anyone that bought sip phones...what online sellers have you used?
03:05.01benjkits an English speaking channel and it is considered a bit rude to use another language that others don't understand
03:05.12ceph__looking at voipsupply.com
03:05.17benjkexcept for throwing in a few phrases
03:05.50*** part/#asterisk alphadad (n=derd@24.83.96.214)
03:07.36benjkJahani: si tu veux parler en francais il faut aller dans une channel privee a cause d'ettiquette
03:08.24wasimmorning benjk
03:08.41benjkhi wasim
03:09.00jahaniok benjk mais le probleme j'ai du male a expliquer en anglais
03:09.19wasimsire, thee were looking for this lowly knave?
03:09.29froguznites
03:09.38benjkwasim: :-D
03:10.27benjkjahani: try us
03:10.55*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
03:11.09benjkwasim: I am still not entirely satisfied that it was right to stop FarFone
03:11.40jahanibenjk je peux vous expliquer en pv?
03:11.48benjkand I think there are still possibilities to do something with it even if you cannot match the low cost of Atcom
03:12.12benjkjahani: yes go ahead -- allez-y
03:12.57jahanithank you
03:13.03benjkwasim: are you listening?
03:13.05wasimbenjk: agreed, there are, but a) no funds b) no direction/project ... thats why the project is dormant until we can find a niche to fit it in, whether is secure encrypted sets, or niche product sets like for the stock exchange of 911
03:13.47wasimor some other API, like voip enabled credit card swipe machines
03:13.57wasimand ATMs etc
03:14.34benjkI am not so much thinking niche
03:15.18wasimor with alternate access like wimax or phs or dect or what have you, like we were initally thinking
03:15.58wasimbut again the problem was the chip makers don't want to talk to us, they want to talk to cisco and the like, so we don't even get datasheets, much less samples
03:16.15benjkwith the PHS/DECT it will need some funding and we need to have a prottype first
03:16.27benjkso that may better be step 2
03:16.54*** part/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com)
03:17.02benjkNow, I know that Apple has interest in IP phones, IF ....
03:17.18benjkthey use Bonjour (aka Zeroconf)
03:17.51benjkand I think it would be far less cost to do a prototype/demo for that than DECT/PHS
03:18.14benjkof course Apple wont like the design of your casing :-)
03:18.17wasimpossibly, yes ...
03:18.26benjkbut that doesn't matter
03:18.54wasimbut less cost != $0 ... which is the main issue, since we've busted the bank on this, and getting further funding in house is problematic
03:18.58benjkwe tell them it is only to cover up the electronics inside
03:19.04benjk;-)
03:19.12wasimwe've made 50 prototypes though ... 30 are in EU, 20 with me
03:19.37benjkBonjour/Zeroconf is a software only thing
03:19.49benjkall it takes is firmware coding
03:19.52mog_homeit is pretty hot
03:19.52[TK]D-FenderBesides.... SIP is anything except ZeroConf :)
03:19.55mog_homezeroconf that is
03:20.04benjkNONSENSE
03:20.09[TK]D-Fender* is like EVERYTHING CONF
03:20.16mog_homelol
03:20.19benjkWe have an RFC 2782 entry for SIP
03:20.27[TK]D-FenderC'mon, how many files do we use now?
03:20.29[TK]D-FenderREALLY
03:20.53[TK]D-Fender:O
03:20.57benjkand we do Zeroconf on Asterisk, advewrtising SIP and IAX
03:21.15[TK]D-Fender100K to go for slackware upgrade!
03:21.30benjkhttp://www.astmasters.net/projects.html#zeroconf
03:21.52_DAWdoes anyone here use the page cmd.  I am trying to determine if I can use alert_info with it..
03:22.08benjkwasim: still there?
03:22.54Pikorowow.. what am I going to do with all these bananas?
03:23.00wasimbenjk: oui, monsieur
03:23.23Pikorogot like 500 of em
03:23.32Pikoroi can't give them away fsst enough
03:23.37wasimbanana nut bread
03:23.44wasimit'll keep for a little while
03:23.59Pikoroyah
03:24.06Pikorodamn banana trees
03:24.17benjkfound a banana republic!
03:24.32wasimPikoro: we're putting different sorts, so hopefully they'll mature staggered
03:24.48benjkanyway, wasim, do you think you can add zeroconf to your firmware?
03:24.50Pikoroyah
03:24.57Pikoromine are staggered
03:25.03Pikorobut there's still too many
03:25.11Pikorolike 25 hands worth
03:25.12Pikoroheh
03:25.29Pikoro20-25 bananas on a hand
03:25.41wasimbenjk: at this point we are not doing any development, to wipe off the cobwebs, get everybody together again would require a formal concrete project of sorts
03:25.46Pikoroi think i am going to get potassium poising
03:26.11wasimbenjk: getting it in is not the problem, convincing management to do so will be
03:26.11*** join/#asterisk [hC] (n=hardcore@209.153.195.139)
03:26.44benjkWasim: thats a real shame, because 1) doing this is fairly straightforward -- not such a big thing and
03:27.38wasimbenjk: agreed that its not that difficult, reinstating the project just to do zeroconf without an actual delivery milestone, or a project in hand is more difficult
03:28.04benjk2) Apple's Bonjour/Zeroconf tsar (who I know personally) has promised that the first manufacturer of an IP phone with Zeroconf will be promoted by Apple on all theior Zeroconf presentations etc
03:28.35*** join/#asterisk [hC] (n=hardcore@209.153.195.139)
03:29.13benjkso if then Apple  themsleves dont want to take it on or buy it from you, at least you get exposure and that should yield some project elsewhere
03:29.35wasimperhaps, perhaps ...
03:29.52wasimi'll float the topic at the dev session tomorrow and see the response
03:30.34benjkand if Apple says "Ok this is a nice proof of concept, so can you do a WiFi phone for us?"
03:31.02benjkthen yuo can complain about not getting any information/support from chip manufacturers
03:31.19wasimbenjk: if Apple says that, I'll marry the lead developers younger sister to any of Steve Jobs's sons
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03:31.27benjkI guarantee you that Apple will open all those doors for you
03:31.43FgL1986hi everyone
03:32.03benjkwasim: that may not be such a good idea because they may be turned off by that ;-)
03:32.06FgL1986i have a question, is there anyome who can tell me a really good web page of asterisk but in spanish?
03:32.29wasimbenjk: you never know, whiteys have a penchant for eastern women
03:33.18benjkthat's why I said "may"
03:34.41benjkanyway, I met Apple's Zeroconf tsar here in Tokyo last year and I showed him a few VoIP gadgets
03:34.49benjkincluding some ATAs
03:34.59benjkHe was very excited
03:35.19benjkand he said "This absolutely neeeed Rendezvous!"
03:35.43benjkback then it was still called rendezvous (before settling the trademark dispute)
03:36.32benjkAnd now that we have got Zeroconf support for Asterisk, half the work is done
03:36.33[TK]D-Fenderok, update about to begin mass install.  BBIAB
03:36.51mog_homezeroconf only works when everything is on same network though benjk
03:37.01benjkall it needs now is some client device with client support
03:37.04mog_homeand i dont seem to have res_zeroconf on my box...
03:37.10benjkmog: not naymore
03:37.23mog_homewhen did it get commited?
03:37.31benjkApple has released Wide Area Bonjour with Tiger
03:37.31mog_homeand it only works when things are on the lan i thought?
03:37.39mog_homehow the hell will that work
03:37.46mog_homeif everyone is broadcasting over the internet
03:38.05benjknot broadcasting on the WAN
03:38.40benjkAnyway, Wasim, think about this and talk to whoever makes the decisions about it
03:38.45wasimbenjk: ok, wilco
03:39.28benjkif it only gets you the exposure from Apple's demos, that should be worthwhile doing it ;-)
03:40.12benjkmog: it works through support in the NAT router and DNS server
03:40.24*** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239)
03:40.36mog_homeokies
03:40.46mog_homebut explain to me how it works
03:40.52mog_homei have my sip device i flip it on
03:40.53[TK]D-FenderWhee, just passed the BitchX upgrade, only 200 package ot go!
03:40.55mog_homeon the other network
03:40.59mog_homehow do i see sip server
03:41.00benjkbasically some sort of fully automated DynamicDNS where all parts play together
03:41.34mog_homeim sorry i must have fallen of the stupid truck
03:41.42mog_homei know how rendevous broadcasts on the lan
03:41.44*** join/#asterisk test34 (i=1000@unaffiliated/test34)
03:41.47mog_homeit pushes it to me
03:41.51mog_homeand thats the point
03:41.58mog_home<PROTECTED>
03:42.05mog_homei would have to poll it no?
03:42.18benjkyou can have the local NAT router pass on your new coordinates and let it rebroadcast that to your hom network's LAN
03:42.47mog_homebut that is configuration... doesnt that defeat purpose
03:42.59benjkso even if you are out of the LAN, other devices on your home LAN will still find you
03:43.05*** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
03:43.06mog_homeyeah
03:43.36mog_homei dont know benjk, it rules on the lan still dont see how it ever is really valuable in the whole wild internet
03:43.45mog_homebut it is cool that they can link lans together
03:46.05tzafrir_laptopbenjk, how useful is zeroconf in a windows-centric network?
03:46.26mog_homethere is apple client for it
03:46.29mog_homeand it doesnt matter
03:46.38mog_homeas you broadcast from linux box
03:46.47mog_homethe software just have to know to look at the info
03:46.52mog_homeso the sip phone supports it
03:46.53mog_homewill work
03:46.59tzafrir_laptopit won't be installed by default. Does it require admin privs to install?
03:47.29mog_homeno tzanfrir
03:47.39mog_homezeroconf is a fancy way of saying
03:47.48mog_homeeach service sends broacast packets out
03:47.51tzafrir_laptopIs there a simple "browser" that I can include in an executable I can put on the (linux) CD that won't even require a windows installer? (like putty)
03:47.55mog_homethat say hey thats me im an apache server
03:48.14mog_homenot that i know of
03:48.24mog_homebut for example some things just work with it
03:54.20*** part/#asterisk Junbug (i=Junbug@69.0.31.27)
03:55.55tainted_can anyone get me a good deal on polycom 301/501?
03:57.12[TK]D-Fendertainted_ : There was a guy unloading 501's in here how spams his msg every once and a while.
03:57.20tainted_lol
03:57.40[TK]D-FenderAlthough I don't really see thepoint of the 301.  No speakerphone or any special features to speak of...
03:57.51benjktzafrir: Zeroconf works fine on Windows
03:57.54tainted_basic callcenter phone
03:58.10benjkmost network printers use it
03:58.10[TK]D-FenderAim for something cheaper :)
03:58.46file[laptop][TK]D-Fender: O.O
03:58.46tainted_[TK]D-Fender like what?
03:58.46[TK]D-FenderSPA-941 wins my vote :)
03:58.46tainted_I don't want playskool garbage tho
03:58.46[TK]D-Fenderit isn't
03:58.46tainted_i can get 301 for 115
03:59.03[TK]D-FenderHmmm.... there is the Uniden UIP-200, but it feels funny
03:59.25tainted_funny?
03:59.27[TK]D-FenderI guess if you just want a MINIMAL phone the 301 would probably be the best feel for the $
03:59.28tainted_as in light?
03:59.31tainted_or as in texture
03:59.46tainted_[TK]D-Fender do u know if it has xml browser?
04:00.14*** join/#asterisk g0mb0 (n=test@external.micom.mng.net)
04:00.19[TK]D-FenderUIP-200 feels.. I dunno... plasticy... hard to describe.  Doesn't use MGCP dial strings,  boots off TFTP almost exclusively, is a little boring looking... and the speakerphone is lacking
04:00.29[TK]D-Fenderbut it has PoE and does look "tight)
04:00.47[TK]D-FenderNOPE.  You'll want an IP 600 for the browser :)
04:01.00tainted_uip-200 does not look good
04:01.16tainted_rounded edges on business phones just don't look right
04:01.20[TK]D-Fendernot really.  It is "solid" though
04:01.20tainted_for some odd reason
04:01.20JunK-Yuip-200 looks like a 30$ analog phone.
04:01.46[TK]D-Fenderheh, owning 2 I'd say "not far" but its better than that.
04:02.32[TK]D-FenderLooks like the 301 is your best bet for a basic phone.
04:02.51JunK-Ytk: when do we planify our sphinx2-redbull night?
04:02.52tainted_any luck with aastra?
04:03.02Pikoroi got an aastra phone here
04:03.06Pikoroit is odd
04:03.09[TK]D-Fenderall of my office workers use IP 600's and I have restricted my CSR's on theirs beacuse of queue call concurrency issues
04:03.24Pikoroif i do a sip reload, it continuously re-registers like once a second
04:03.24[TK]D-FenderJunK-Y : booyah!
04:03.31[TK]D-FenderDid you go tonight?
04:03.39JunK-Ysure.
04:03.45[TK]D-FenderHow was it?
04:03.50[TK]D-Fenderanything special?
04:03.58benjkACT P104 is a solid business phone at a low cost
04:03.59JunK-Ynope
04:05.01tainted_ACT P104?
04:05.19benjkit won't win any avantgarde design competition -- its very conservative office design
04:05.28*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:05.34benjkbut it is a very solid phone
04:05.53JunK-Ytk: check that for the sphinx2, maybe we could work on that one night.
04:06.04benjkand has 10 lines with control lights that show you if you have parked somebody some channel
04:06.26benjkThey even have IAX firmware for it
04:06.58benjkbut those silly Taiwanese never released the firmware for real
04:07.17tainted_silly taiwanese!!
04:07.41benjktainted: Advantage Century Communications -- something like that
04:08.02benjkthey mostly sell to companies who rebrand it
04:08.17benjkso you can find it under all sorts of names
04:08.26benjkbut its usually called P104
04:08.45tainted_have u had any luck with the yuxin?
04:09.11tainted_damn these bitches!
04:09.14benjkHere in Japan one of the big Japanese names use it and sell it against the Cisco 7940 under theri own brnad
04:09.15tainted_charge me 20$ in handling
04:09.36tainted_the yuxin? or the aastra?
04:09.43benjkFujitsu or HItachi, I think, dont remember which one
04:09.52benjkno the ACT P104
04:15.06Pikorothere a doc out there on spandsp and asterisk?
04:15.12Pikoroi gotta get this caller id stuff working
04:17.21[TK]D-FenderPikoro : Check the Wiki, there are lost of links on that specific topic
04:17.24[TK]D-Fender~docs
04:17.26jbotdocs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
04:17.35Pikoroyah, all i see is patching for fax use
04:17.51[TK]D-Fenderthere are real guides for SpanDSP there.
04:17.55[TK]D-FenderI've seen them
04:18.04Pikorok
04:18.29[TK]D-Fenderwell, I'm off for the night.  Later all
04:18.34Pikorolater
04:18.51*** join/#asterisk jontow (n=jontow@secure.bsd.st)
04:29.23*** join/#asterisk SplasPood (i=nobody@paravolve.net)
04:30.55*** join/#asterisk Entegrity (n=Entegrit@c-24-34-120-110.hsd1.ma.comcast.net)
04:31.25*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
04:46.17Pikorohmm... do grandstream phones (gxp-2000) support adsi?
04:46.27kuku5does a2billing work with 1.2?
04:46.36mog_home<PROTECTED>
04:46.40*** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239)
04:46.41mog_homethat is  a sip phone right
04:46.46Pikoroyah
04:46.56mog_homethen its not an adsi phone
04:47.06mog_homean adsi phone is a super analog phone
04:47.43[TK]D-FenderYay, actually running * at home for the first time in 5 months!
04:48.31hypa7iabut not *@h
04:48.32hypa7iahehe
04:48.42[TK]D-FenderNEVER A@H!  Ewww!
04:48.46[TK]D-FenderCLI goodness!
04:48.47Pikoroi need to find a way to get this callerid working
04:49.38Pikorooh you know a@H 0wnz j00 fender ;p
04:50.19[TK]D-FenderNah... I'm public domain :O
04:50.40[TK]D-FenderOwn THIS! Hi-yaaaaaaaaa!
04:51.10Pikoroif i build from CVS-HEAD is it going to be 1.3 now?
04:52.53Pikorodamnit.. without callerid support, we're gonna tear down this asterisk box
04:52.59Pikorothis kinda sucks
04:53.05[TK]D-Fenderdon't give up hope, start Googling
04:53.09*** join/#asterisk jahani2 (n=k@adsl-186-44-192-81.adsl.iam.net.ma)
04:53.16asterboyWho has taken the polycom certification?
04:53.17Pikoroi've been googling for like, 4 days now
04:53.17Pikoroheh
04:53.26Pikoroi got polygraph certification
04:53.44asterboysqueeze those but cheeks everytime they ask a question?
04:54.00asterboyor breath irregular?
04:54.40asterboyso I call an equipment vendor to buy some polycom IP601 phones.
04:54.47asterboy"Do you have certification?"
04:54.51asterboyNo
04:54.59asterboy"Then we can't sell them to you"
04:55.18asterboyGreat, I'll take my order for 50 units to ebay...now saw off.
04:55.42asterboywhat a joke...certification for a phone...give me a break!
04:56.10asterboyDoes Cisco do this with their phones?
04:56.15orlockhmm
04:56.20[TK]D-FenderPikoro, Think this may apply to you  http://www.voip-info.org/wiki/view/Asterisk+bounty+NTT+CLID
04:56.26orlockcretification for phone gear itself is actually not unreasonable
04:56.29mog_homeyeah i think they do asterboy
04:56.39mog_homecisco makes it hard for you to buy their shit
04:56.41orlockbut not for stuff thats going into a data network i would think
04:56.42mog_homeit makes no sense
04:56.45mog_homejust sell us stuff
04:56.48mog_homewe will buy ity
04:56.53orlocktelco's dont like it when you plug in gear that cooks things
04:56.53asterboyIt's not like your WalMart shopper is looking for an IP phone.
04:56.56mog_homei never understood all this crap
04:56.59orlockbut that shouldent stop you buying it
04:57.19benjkHey, that one was put there by myself
04:57.30benjkthe NTT CLID
04:57.46[TK]D-FenderOh, you ARE back :)
04:57.51asterboyya, I'll get around it, but its annoying..
04:58.09benjkI was in the private chat with that French speaking fella
04:58.09[TK]D-FenderPikoro here has been having problems, and I guess there ISN'T a real solution yet
04:58.28benjkneeded to give him a roundup on Asterisk
04:58.42asterboybenjk, thats what you were talkin about last night...the bounty
04:58.43[TK]D-FenderC'est toujours la faute des maudites Anglais, non? ;)
04:58.50benjkwould have annoyed the channel to have French stuff in here
04:58.51mog_homeyuck
04:59.08benjk:-D
04:59.12[TK]D-Fender;)
05:00.32benjkyeah the Japanese caller ID is funny
05:01.14benjkthe FXO interface has to pick up the line, listen to the caller id, then hang up to let the call setup continue
05:01.49benjkThis is ojne of those things were I Digium is bad news for Asterisk
05:01.51asterboyhang up?
05:02.01asterboydoh
05:02.04mog_homewhat benjk?
05:02.17benjkThere is a Japane4se caller ID decoder released under LGPL
05:02.37benjkit could be copy pasted into zaptel.c or wcfxo
05:03.05mog_homeokay?
05:03.10benjkthe licenseing is even fine for commercial distors if it is orgnaised into a library and dynamically linked to
05:03.20benjkbut Mark refuses to use it
05:03.25[TK]D-FenderStill sounds messy....
05:03.29mog_homeumm okay?
05:03.29benjkhe wants it disclaimed
05:03.38mog_homeare you sure he just doesnt want it?
05:03.43mog_homeor that its good code?
05:03.48[TK]D-FenderYeah, Mark likes to be able to own everything :)
05:03.56mog_homeand if its something easy just go and reinvent it
05:04.04benjkno he said if Voicetronix disclaim it then he will use it
05:04.13benjkLe me ask you this ...
05:04.20mog_homebah fender
05:04.24[TK]D-FenderPolitics SUCK.....
05:04.34benjkdid Digium get a disclaimer of Linux libraries which are LGPL?
05:04.35mog_homemark just doesnt like being cought with his pants down
05:04.56mog_homeben there are several lgpl code that got into asterisk
05:05.01benjklike say glibc or stuff like that
05:05.02mog_homelike the stuff i am working on
05:05.06benjkEXACTLY
05:05.07mog_homeuses an lgpl library
05:05.22benjkso why not an LGPL Japanese caller ID decoder?
05:05.26mog_homemy question is though
05:05.30mog_homedid you write the code
05:05.37mog_homeor just say you could add this library in
05:05.40mog_homeand bam it will work
05:06.28mog_home?
05:07.15benjkThe code that still needs to be written is the part where the FXO goes OFF-HOOK before reading the FSK encoded signal from the CO and then go ON-HOOK again
05:07.27mog_homeif you wrote that code
05:07.30mog_homeand submitted it
05:07.36mog_homeand he rejected it
05:07.39mog_homeitd be different
05:07.49benjktwo engineers I have asked about this and shown the code said it would be about two days work
05:07.52mog_homejust saying woohoo i found this lgpl code
05:07.56mog_homelets do it
05:08.16benjkthe code works perfectly well with the Voicetronix card and Asterisk
05:08.36benjkjust not with Zaptel
05:08.52benjkand Zaptel is more or less Digium's baby now
05:09.39benjkso it wouldn't be too much to ask Digium to use that LGPLed code from Voicetronix and do the off-hook on-hhok thingie for Zaptel
05:11.31kramso where's this code that i apparently don't wnat?
05:12.11kramerr want?
05:12.36benjkits in Voicetronix' driver
05:12.56benjkLGPLed
05:13.02kramokay you have a link?
05:13.15benjkyup just a second
05:14.03benjkhttp://www.voicetronix.com.au/Downloads/vpb-driver-2.4.9.tar.gz
05:14.28benjkthe decoder is in jp_cid.cpp in directory src
05:14.40mog_homec++ ew
05:15.08kramew, it's c++
05:15.16mog_homehey i beat you too that
05:15.34kramyou mean cid_jp.cpp ;-)
05:16.15benjkunfortunately yes, but then again, what wouldn't you put up with for Japanese caller id support if you were in Japan ;-)
05:16.21benjkoops
05:17.36[TK]D-Fenderfile, S518 is running 100%, My * server is back up, my new SPA-941 is working like a charm.  once I get my SPA-3000 set up its all GOLD!
05:17.44kramhrm, we already have most of this in asterisk
05:17.44bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
05:18.00h3x\kb bbz spammer
05:18.08mog_homeyay! asterisk
05:18.10krami wonder if we're just not oding it right
05:18.18h3xbbz: try paypal
05:18.19h3xer
05:18.21h3xeBay
05:18.30benjkanyway, the English documentation of NTT's line interfaces incl the clid stuff is at
05:18.46benjkwww.astmasters.net/stuff/NTT-TSI-English-Ed5.pdf
05:19.22benjkMark, you need to get the FXO interface to go OFF-HOOK to read the Japanese caller id from the CO
05:20.02benjkand once you've captured it, it needs to go ON-HOOK again to let the call setup continue
05:20.16file[laptop][TK]D-Fender: beautiful
05:20.16krami probably need to get a dump of it
05:20.20benjkA bit silly, but thats the Japanese for you
05:20.35benjkthey like to mess with things to be different
05:21.02kramis this number only?
05:21.12benjkI can let you log in to a box with an NTT line connected
05:21.19benjkI think so yes
05:21.23[TK]D-Fenderbenjk, Parts of New York are like that too... difference CallerID spec from the rest of Northa America...
05:21.32kramyou think so?  you haven't used it?
05:21.33benjkbut as I said, this is in that PDF document
05:21.57benjkwell, I get caller ID on a Japanese phone only as numbers
05:22.27benjkbut, I couldn't say that there is no service available that may provide caller ID names
05:22.42benjkI doubt it but I cant say for sure
05:22.58kramyah this is number only at least what's described here
05:23.03benjkThe service is officially marketed as Nambah-disupleh
05:23.11benjk=> number display
05:23.21CpuIDkram, you able to complete g729 codec orders? :)
05:23.27benjkso I guess it means that its only numbers
05:23.27kramno, i'm not
05:23.28CpuIDbeen waiting 2 days for someone to fill one lol
05:23.34mog_homelol
05:23.38mog_homewhy not kram
05:23.51CpuIDsent a mail to customer service bout it, might need to give them a poke, just letting you know :)
05:24.04mog_homewell its turkey day
05:24.10mog_homeand no one is at office
05:24.42benjkD-Fender: that sucks!
05:25.50[TK]D-FenderNot for me.. I'm in a greener land :)  Actualy kinda white right now, but SEMANTICS!
05:26.22CpuIDhehe point :)
05:26.24CpuIDforgot bout that
05:26.36CpuIDthats ok, didnt mean to be pushy kram just givin some feedback :)
05:26.37mog_homeits not completely automated yet
05:26.41mog_homeyeah
05:26.43CpuIDwhens the office open again anyways?
05:26.48mog_homemonday
05:26.57CpuIDah k coo, im in AU so we dont have turkey day lol
05:27.01mog_homesorries
05:27.04CpuIDhehe
05:27.06mog_homeyou should do it anyways
05:27.08mog_homebe different
05:27.35mog_homewhats an australian holiday that is only celebrated there
05:27.36benjkCpuID: That would be Turkey with beetroot then ;-)
05:27.36file[laptop]mog_home:   :D
05:27.38mog_homeill start it here
05:28.03benjkANZAC day
05:28.10mog_homeanzac?
05:28.16benjkANZAC
05:28.18mog_homeexcuse my ignorance
05:28.34benjkAussie and NZ Army Corps
05:28.48benjkWWI and WWII veterans
05:28.53mog_homeoh cool
05:28.56mog_homewhat day is that
05:29.05benjkGalipoli rings a bell?
05:29.26benjkWWI battle in the dardanelles straight?
05:29.38file[laptop]nini all
05:29.49benjkMany Aussies and Kiwis were wasted there
05:29.55kramokay i'll take a look at this
05:30.03benjkthat's a big thing in OZ
05:30.18mog_homeno what do you say benjk....
05:30.24benjkMark, thanks a lot
05:30.33mog_homeaww
05:30.37mog_homethats sweet
05:30.39file[laptop]that was nice, you  get a muffin
05:30.54kramyou can thank me if it works
05:30.55benjkI appreciate it and I have to say it seems things have changed for the better
05:30.58kramor if i make it work
05:31.16file[laptop]kram: you should go to sleep like I am
05:31.16kramand if i don't, then just go about your business bad mouthing me and digium as usual
05:31.17benjkI thank you just for making the effort
05:31.20{zombie}yeah, anzac day is a big thing here (25th April) - and Australia Day (26 Jan)
05:31.30kramheh
05:31.52mog_homeapril 25th
05:31.56mog_homemark it mark im taking it off
05:32.03benjkI amy criticise but I am fair
05:32.09mog_homeso i can stand with my australlian forefathers
05:32.18benjkI give credit when and where credit is due
05:32.32kramooh maybe i need to find v.23
05:33.42CpuIDlol yea anzac day
05:33.43CpuIDthats us
05:34.25benjkmog: if you want to honour your Aussie forefathers, you gotta have an Aussie burger
05:34.32mog_hometrue
05:34.33benjkwith beetroot
05:34.38mog_homeill go to outback that day lol
05:34.41benjkyummy
05:35.05[TK]D-Fender*yawn* well its dfinately quitting time for me.... back again tomorrow
05:35.41[TK]D-Fenderlater all
05:39.09CpuIDlol australian forefathers
05:40.04*** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc)
05:40.29{zombie}as in the convicts?
05:40.36mog_homesure
05:40.52mog_homeactually my family has history of intense violence from generation to generation
05:41.11{zombie}it's funny how that was something nobody wanted to admit to - being descended from the convicets, but being descended from someone on the "first fleet" is apparently something to brag about now
05:41.34mog_homeheh
05:43.19*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
05:45.58*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
05:46.05benjkyeah but in those days to become a convict could well just have meant you were poor
05:46.47mog_homeor dumb
05:47.33Kattyor an embarassement
05:47.47benjkKatty: LOL
05:48.55KattyzomgwtfbbqlolzkthxbiNEXT
05:50.06benjkdid you fall asleep at the keyboard?
05:50.21Kattyheh
05:51.51Pikoroqwertyitis
05:59.03*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
06:09.20*** join/#asterisk clive- (n=pirch@ndn-165-156-99.telkomadsl.co.za)
06:13.46*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
06:16.14*** join/#asterisk ComputerWarm (n=workingg@66.244.235.222)
06:16.38ComputerWarmEvening all; what is the recommended billing software for asterisk? i am running php/mysql 5
06:17.58alephcomuh, oh....  Flameware material.... :-)
06:18.57KattyComputerWarm: i don't even use billling software, so i dunno
06:21.03alephcomComputerWarm: I use opensource software that I modified from ASTCC.  The newest version hardly represents ASTCC anymore but that's where it started.  www.aleph-com.net/astpp
06:21.10alephcomIt all depends what you want though.
06:22.30asterboyI'm checking out astbill: http://astbill.com/whatis
06:22.36asterboylooks like a good choice.
06:22.39ComputerWarmalephcom i am just looking for something that people can use prepaid minutes and when the money runs out its cuts them.
06:22.43camonzhow so...., by example if i wanted to create an interface so it could constantly notify to a lcd screen the current time of the call and the price of it
06:22.44asterboyDrupal is a requirement though
06:22.52camonzhow would i interface it with mysql
06:23.22ComputerWarmasterboy i tried astbill and i got sick of rewriting the script. i would on it for like two weeks.
06:23.27ComputerWarmwould==worked
06:23.38alephcomComputerWarm:  We can do that easily.  The requirements are fairly minimal.  MySQL, apache, and a couple of perl libraries.
06:23.39asterboyglad you warned me.
06:23.51asterboyI was going to try it...but Drupal turned me off
06:24.09alephcomI'm going to have an oscommerce plugin shortly so users can purchase credit through an oscommerce store.
06:24.17alephcomThe developer is working on it.
06:24.38ComputerWarmalephcom anyidea when it will be ready?
06:24.59asterboyI just want a SIMPLE script to parse the CDR info and report it in a nice clean html page.
06:25.11mog_homeastbill
06:25.26asterboyya, but ComputerWarm says its a script nightmare.
06:25.26alephcomComputerWarm:  should be ready next week.
06:26.01ComputerWarmasterboy the reason i am having the problem is because i use php5/mysql5 i believe
06:26.31asterboywell it says it want mysql5...php5 can be another headache.
06:27.54asterboyI was not at all impressed with Drupal...not sure where they are at now.
06:29.15*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
06:31.27ComputerWarmquestions with asterisk can you get someone to say there name if there callerid is unknown?
06:31.36ComputerWarmto speak it and let me hear it?
06:31.46alephcomJeremy @ Nufone would definitely say to write your own application.
06:32.07alephcomYes you can, I'll go have a look and find it.  I'm sure I found functionality like that somewhere.
06:32.10mog_homewhat do you mean computerwarm
06:32.16mog_homelet user say it
06:32.22mog_homeand then play you back recording
06:32.29ComputerWarmmog_home yes
06:32.51mog_homethere is a 20 line dial plan thing on voip-info.org
06:33.01mog_homeits like call screen or something like that
06:33.03mog_homeits very cool
06:33.21mog_homechris hozian at digium wrote it, if you dont find it ill track him down to send it to you
06:33.41ComputerWarmok thanks i will look through i guess the different extensions.conf files?
06:34.21*** join/#asterisk zobia (n=laura_sh@218.6.242.212)
06:34.28zobiahello everyone
06:34.42asterboyhowdy
06:34.51mog_homeits just some macro
06:34.57mog_homebut it does all that for ya
06:35.05alephcomhttp://www.voip-info.org/wiki-Asterisk+cmd+Dial   You would have to do a some testing to check on callerid.
06:35.30zobiai send out a call to asterisl spool. is there any way i can know the call hangup or not and hangup reason?
06:36.22alephcomTake care everyone
06:36.35ComputerWarmoh he left
06:38.48zobiabenjk , are u there?
06:42.03*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
06:43.02*** join/#asterisk oriontkn (n=oriontkn@fullerton-cuda-2-70-37-76-198.lmdaca.adelphia.net)
06:43.13oriontknhi
06:44.35asterboyhi
06:44.43benjkyes I am here
06:46.23oriontknI have a n00b asterisk question.. I'm getting registration rejected trying to connect iax to fwd... I enabled iax in my profile and my username and pw are correct.. is it me? or is something going on with fwd?
06:48.02benjkFWD sometimes suffers from split memory syndrome
06:48.25oriontkniax and sip reg database doesnt match?
06:48.25benjkthe SIP side doesn't know what the IAX side is up to and vice versa
06:49.33oriontknhmm....  that sucks
06:49.33benjknow, as to your problem, it often can be fixed by disabling IAX then wait some time, at least 10 minutes, then enabling it back again and wait again
06:49.46oriontknok
06:49.46*** join/#asterisk oej (n=Olle@apollo.webway.se)
06:49.52zobiabenjk. if i send out the call by the call file , is there any way i can get a hangup reason code?
06:50.13benjktake into account that FWD's IAX service is still experimental
06:50.33benjkzobia, I honestly don't know
06:50.48zobiaokay. no problem.
06:51.20oriontknyea.. I was having problems with sip being a double nat... so I thought IAX might have been the answer but I have read in the forums it has been up and down.. but the last I saw nov 10th it was "stable"
06:51.36oriontknthanks for the advice.. I'm going to try the disable/enable
06:51.49oriontkndo you use ISDN with asterisk?
06:52.16benjkWell, the SIP based service tends to be flaky as well
06:52.37benjkwhat do you intend to use FWD for?
06:53.06oriontknjust as free sip to friends that also have fwd accounts
06:54.20Dr_Raywhy not just bypass FWD
06:54.24oriontkncurrently I have a sipurs spa3000 that regs to it and that works fine.. but my friend gave me a Cobalt Raq4 to play with.. so I setup asterisk on that and pointed the spa3000 as an asterisk extension and got that working...
06:54.50benjkso you and him are both behind NAT?
06:55.02oriontknI just wanted to see if I could get fwd working with asterisk
06:55.13oriontknyes, both nat
06:55.18oriontknmost of my fiends are nat and dhcp
06:55.26oriontknI'm double nat and dhcp at the moment
06:55.33benjkand he doesn't have Asteriks?
06:55.43oriontknnot yet
06:55.59oriontknI think I want to give him the raq4 with asterisk on it for his switch and then set one up for myself
06:56.12oriontknI currently have an ISDN BRI circuit as my home voice connection
06:56.21benjkok
06:56.31oriontknI want to run the BRI into asterisk and then use some hfc cards to use my isdn phones as stations
06:56.48oriontknbut it seams NT mode support of ISDN sets needs some more development
06:57.15benjkI am not all too familiar with BRIstuff
06:57.23benjkI am not in Europe
06:57.28oriontknI am in the US
06:57.35benjkI am in Japan
06:57.38oriontknoh
06:57.38oriontkncool
06:57.51benjkJapanese BRI is yet again different
06:58.02benjkno support and no cards
06:58.07*** join/#asterisk newmember (n=newmembe@70.72.189.149)
06:58.07oriontknso are you suggesting ditching fwd in favor of asterisk to asterisk.. or something like iaxtel?
06:58.13oriontkn;(
06:58.14oriontknthat sucks
06:58.17benjkand it is rabidly dying
06:58.20oriontknI'm sorry to hear that
06:58.24benjker rapidly
06:58.29oriontknISDN is such a great technology that never was
06:58.43oriontknI hate packet switched voice... when you can have circuit switched voice
06:58.51benjkwe have 26Kbps VDSL and 100Mbit FTTH over here
06:58.52oriontknwhich is quickly becomming a dying breed
06:59.00benjkso nobosy wants ISDN BRI
06:59.11newmemberfrom CLI how do I make a Digium FXO interface hang up
06:59.12oriontknyeah
06:59.28h3xwhat never happened was ATM
06:59.47benjkas to FWD versus IAXtel
06:59.48oriontknnewmember... I think there is a 'soft hangup' in the help
06:59.54oriontknI'm not sure I've never done it
06:59.59benjkIAXtel is also not very reliable
07:00.09h3xthey shoulda just made the fixed atm cell size bigger than 53 bytes
07:00.13benjkyes softhangup
07:00.13h3xlike
07:00.19h3x1k or something
07:00.26benjklike 54 bytes!
07:00.30benjk:-)
07:01.05oriontknlike ISDN and DSL I think wide area ethernet has quashed ATM to some degree
07:01.21benjkAsterisk to Asterisk via IAX would be a good thing to do
07:01.22h3xbut atm AAL5 could have done variable bit rate, compressed voice
07:01.48h3xsome cable providers use it for voice
07:01.49h3xlike coz
07:01.50h3xcox
07:02.02benjkcogh
07:02.08benjkcough
07:02.14h3xcocks
07:02.28benjkcoughs
07:03.01benjkorion, what's your router there
07:04.14newmemberSo I trying soft hangup with this:  This does nt work, but I am close?  soft hangup zap/1
07:04.32oriontkndoing NAT?
07:04.39oriontkncrappy linksys RT31P2
07:04.41benjkyes
07:04.52benjkcan you replace it?
07:04.55oriontknthen behind that is a Netgear RT311
07:04.58oriontknyes
07:05.31benjkget yourself an old vintage PC, slim minitower if possible
07:05.48benjkand get a copy of Wolverine
07:06.20benjkhttp://www.coyotelinux.com/products.php?Product=wolverine
07:06.47benjkreplace the hard disk with a CompactFlash IDE adapter and 32MB CF card
07:06.59benjkstick two additional NICs into the box
07:07.14benjkthen put Asterisk in the hardware DMZ
07:07.40benjkthis way, you get your Asterisk box to be on the global IP without any NAT
07:07.58benjkand you still have your NATed LAN
07:08.23benjkand Wolverine does IPsec too
07:08.38benjkso you can tunnel out to your friend
07:09.49oriontknI could stick wolverine on the cobalt raq4
07:09.58benjkNo
07:10.04oriontknno?
07:10.15benjkIts a special embedded distro
07:10.21oriontknmake asterisk another box and make the raq4 my nat/vpn
07:10.31benjkI doubt it will run on the raq
07:10.34oriontkno
07:10.45benjkof course you can try
07:11.04oriontknyea... it was a bitch getting asterisk to run on it...
07:11.23benjkit only needs a Pentium 75MHz and 64MB RAM, 32MB Hard disk (CompactFlash) though
07:11.38oriontknsolaris decided to just not implement a lot of basic bsd style calls and telephony objects.... so I had to modify the source of asterisk to get it to run
07:11.52benjkwould be a bit of a waste to use the raq for that
07:11.52oriontknhave you seen the soakorus (sp?) router boxes?
07:12.13benjksoekris?
07:12.20oriontknyes
07:12.27benjkheard of them
07:12.44oriontknI can never remember how to spell it. I couldn't find the bookmark
07:12.45CpuIDsoekris*
07:12.48CpuIDlol
07:12.54benjkbut after running Wolverine for two years I can say that it is the best IT product every
07:12.57CpuIDreminds me, i need to order a soekris box for my new home ap
07:12.58oriontknI thought about using one of those a while back...
07:13.29benjkyou can buy a hardware Wolverine from that website I posted though
07:13.45benjklittle blue box, looks like a Netgear SOHO router
07:13.50CpuIDpfsense == good
07:13.57benjkbut is equivalent to a Cisco Pix
07:14.12benjkat a tiny fraction of the cost
07:14.27benjkand the support for it is absolutely impeccable
07:15.08benjkyou post to the forum and the guy who makes Wolverine usually responds with something that get you going within less than 24 hours
07:15.23benjkJoshua Jackson is his name
07:15.36benjkhe does nothing else but Wolverine
07:15.42benjkgod bless him
07:15.48oriontknyes
07:16.51benjkand if that all wasn;t nice enough already, it even has a user friendly web interface
07:17.31benjkI have had oodles of issues with routers here in Japan
07:17.56benjkI trioed every product on the market in the price range up to 3000 USD
07:18.05*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
07:18.12benjknothing but trouble with all of them
07:18.35benjkthen I used Wolverine, and never had any problems since
07:18.39*** join/#asterisk escribzz (n=escribzz@wsip-24-249-174-19.ph.ph.cox.net)
07:21.17benjkanyway, this would be a good way to get your Asterisk box out of NAT
07:21.37oriontknyeah
07:21.43escribzzhello
07:21.47oriontknhi
07:21.54escribzzhow are you guys
07:22.06oriontknGood. You?
07:22.22escribzzdoing good just trying to get some things knocked out on a long weekend :)
07:22.36escribzzany exciting topics tonight?
07:23.17benjkknocked our
07:23.20benjkout
07:23.27benjksounds painful
07:23.37oriontknbenkk: so would you recommend just doing IAX between asterisk boxes when mine is no longer natted?
07:24.15benjkeither that, or you just run a server for all your friends
07:24.44*** join/#asterisk [chico] (n=chico@p54914A62.dip.t-dialin.net)
07:26.27[chico]who can help me by a problem??
07:26.42benjkby a problem?
07:26.49benjkor buy a problem?
07:26.52oriontknI will sell you mine!
07:26.54oriontkncheap
07:26.57oriontknclose out prices
07:27.03escribzzI'll give you my problems lol
07:27.07benjkhey I can sell you truckloads of problems
07:27.31[chico]Nov 25 08:27:02 NOTICE[7678]: chan_sip.c:4045 sip_reg_timeout:    -- Registration for 'xxxxxxx@sipgate.de' timed out, trying again
07:27.31[chico]Nov 25 08:27:02 WARNING[7678]: chan_sip.c:1401 create_addr: No such host: sipgate.de
07:27.44oriontknIf you act now I will include all of the problems of everyone I know
07:28.09benjkchico is your dns ok?
07:28.10escribzzprobably sip packet can't get back to the source
07:28.18escribzzprobably a nat issue eh?
07:28.30benjkfirst thing to check is dns
07:28.37benjkthen NAT
07:28.37escribzzya dns good call
07:29.19[chico]could it be a proxy problem?
07:29.33benjkit could be a dns problem
07:29.43escribzzit could be but can u do a ping from the box and it resolve?
07:29.47benjkdig sipgate.de
07:29.53escribzzdig it on the box
07:30.19razuhi
07:31.19razuwhat could be my problem ... i'm trying to put moh from class reklaam and it doesnt go there, it triest o play default class :S
07:31.28razuexten => 111,6,Dial(SIP/111|10|m(reklaam))
07:31.49benjkMoH can be stubborn
07:31.59benjktry restaring asterisk
07:32.02razudid it
07:32.09razutwice :S
07:32.25SERGEUS|WORKhi everybody!
07:32.25benjkkilled the mpg123 processes?
07:32.27[chico]ok thank you i´ll try it
07:32.34SERGEUS|WORKneed a smart advice :)
07:32.53benjkoh, we're short on smart advice today
07:33.03SERGEUS|WORKas always :))
07:33.07razubenjk : i need to kill the mpg123 processes ?
07:34.06SERGEUS|WORKhow to correctly finish APP, in case of error? what is a correct API call for that? shoud i simply call LOCAL_USER_REMOVE(u); return 1; ?
07:34.09benjkrazu: asterisk kicks off mpg123 and they continue to run in the background independently of asterisk
07:34.42benjkso if they survive when you restart asterisk, then they may still be playing your old music
07:34.54ComputerWarmquestion please. i am calling in on a x100p and for some reason asterisk is showing its calling in twice?
07:35.11benjkI remember to have run into that sort of thing before
07:36.50benjkSERGEUS, I think you are better off in #asterisk-dev
07:38.43*** join/#asterisk gvag11 (n=g@ipa77.4.tellas.gr)
07:38.51gvag11hi all
07:38.59*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
07:38.59SERGEUS|WORKbenjk, thanks :)
07:39.14lmehi guys !
07:39.18oriontknhi
07:39.28oriontknHappy Thanks Giving everyone
07:39.33ComputerWarmanyone with any ideas on my problem?
07:39.34gvag11anybody familiar with frame slip and (or) IRQ sharing problems?
07:40.08benjkunfortunatly all too familiar with IRQ sharing
07:40.47benjkbest solution to that problem: buy a Mac
07:40.58benjkno IRQ issues, never
07:41.08benjkuse Yellod Dog Linux
07:41.18benjker Yellow Dog
07:41.20gvag11benjk: I have a problem with cut short faxes(spandsp).. But i have no IRQ missing, libtiff fine, timing fine... Any idea
07:41.40oriontknjust curious...how do Macs handle hardware pci interrupts ?
07:41.58gvag11benjk: can Digium boards work on a Mac ??
07:42.17benjkwell, I am familiar with the IRQ problems but that doesn't mean I have found a cure
07:42.24benjkeverybody has these issues
07:42.27escribzzhas anyone had to ever have to edit thier /etc/security/limits.conf  file on a RH box cause asterisk has too many files open?
07:42.34trixterI dont have those issues
07:42.38benjkyes, Digium board work on the Mac
07:42.46trixterI also rarely max out a box in terms of what it can hold :P
07:42.49benjkuse Yellow Dog Linux
07:42.57trixterand disable much of the crap that causes the problem
07:42.58escribzzget a good box to fix the irq sharing lol
07:43.25benjkyou can run 6 digioum cards in a Mac, no problem
07:43.26escribzzcheap MB's dont handle it like high end intels do
07:43.27gvag11I am working on a Dell SC1425... Considered to be a good machine (i think) ...
07:43.28lmeoriontkn: simple.... nothing interrupt a mac... neither endusers
07:43.38trixternormalyl sharing is caused by a cheap motherboard doing auto assignment in ways it shouldnt..
07:43.45escribzzI've got all tell poweredges 2850's and got rid of the sharing issue
07:44.02escribzzon a cheap abit I had problems all over
07:44.03*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
07:44.06gvag11benjk: and what is a MAC model that i could start working with
07:44.09escribzzcould only run one item at a time
07:44.19escribzzand every other reboot different cards would work
07:44.30oriontknlme: I know this is a little out of context.. but does it use polling then?
07:44.33escribzzanyway anyone ever have to mess with /etc/security/limits?
07:44.35benjkorion: I don';t know how it works, but Macs have traditionally been used by Musicions for realtime/audio
07:45.19lmeoriontkn: i stop using mac after Apple IIe
07:45.19oriontknescribzz: maybe try forcing the PCI IRQ assignment in the bios... update ECSD data and make sure PNP OS is set to NO
07:45.25lme+ped
07:45.26benjkand I think that has something to do with it. They simply had customers who needed an architecture that can handle these things
07:45.34oriontknit was the LCIII and Centrus 610 for me
07:45.45benjkgvag: any PCI Mac is fine
07:45.59gvag11even the mini-MAC?
07:45.59escribzzOriontkn: for the IRQ issue I just bought higher end machines, I learned my lesson fast :)
07:46.07lmebenjk: which mac r u using ? I'm very curious about moving to mac...
07:46.28benjkif it is just for fooling around, get a vintage one, like 9000 series, they are very robust and cheap cause they cant run MacOS X, only Linux
07:46.54benjkthe Mac Mini doesn't have a PCI bus
07:47.03gvag11ok
07:47.14benjkyou can run Asterisk on it, but you cant put any boards into it
07:47.21lmei need to get a robust hardware solution
07:47.23escribzzOriontkn: but now on those higher end machines I'm getting too many open files, Messed with ulimit, messed with /proc/sys/fs/file-max no change so I'm looking at /etc/security/limits.conf, just wondering if anyone else have have that problem
07:47.24razucan i do somekind of certain file playback while a phone rings ? So i dont need to use moh and queue lists
07:47.51benjklme: I am using Xserve, G4 towers, 9000 series, Cube and Powerbooks
07:48.09benjkand my Intel stuff is all IBM and IBM only
07:48.13gvag11Asterisk + Digium might work fine but the whole solution is problematic since it relys on motherboard/cpu and the last one changes everytime and you are never sure unless if you first pay and then find out ......
07:48.24oriontknescripzz: not familiar with the problem.. but have you run an strace on the asterisk process to see what files its opening?
07:48.50lmebenjk: thanks, I'll get one to test !
07:49.00trixterI almost feel like doing more work tonight..  got frozen burritos (cooked of course :) and some alcohol...
07:49.09trixterbut meh I prolly wont do any work cause work sucks
07:49.15implicitsucks
07:49.16escribzzoriontkn: just all of the sip session files and zaptel stuff but its too many
07:49.28oriontknhow many is it?
07:49.31benjklme: if you want to run Asterisk on OSX, use my installer at http://www.sunrise-tel.com
07:49.32implicittrixter, you're back :)
07:49.44implicitbenjk, installers are gay as fuck
07:50.06benjkimplicit, not mine
07:50.13trixterI never left, I was just ignoring you :P
07:50.25benjkthey have tons of testing and q&a in them
07:50.26implicitwhy, i helped you so much that night
07:50.26implicit:)
07:50.29lmebenjk: have you ever tested 1.2 ?
07:50.31zobiais there any where to related a call file with one CDR record?
07:50.46benjkvery briefly yes
07:50.52lme& ? no problems ?
07:51.03implicitlme, you can't prove by counterexample :)
07:51.05benjkbut I am finishing a new distribution with new GUI apps
07:51.26*** join/#asterisk billatq (i=bill@aggienerds.org)
07:51.32benjkso I wont be looking into 1.2 seriously before that is finished and released
07:51.45billatqWhee, I just finished a very simple native mac os x iax client
07:51.55billatqat least makes phone calls at the moment
07:52.11benjkbillatq: Cocoa?
07:52.11zobiabenjk, the same question. but now i don't need the hangup cause code. just wnat to know the call is hangup or not ,is it possible?
07:52.15billatqbenjk: Yeah
07:52.26billatqI just need to start fleshing out stuff
07:52.28trixterI had a weird dream earlier ...  I was playing some video game pack that was a full sized arcade game that was full of bugs and the bugs anoyed me cause it didnt play like the original video games 20 years ago
07:52.31benjkcool
07:52.41billatqI made a cocoa wrapper around iaxclient to use
07:52.50benjkbillatq: do you intend to make this payware or free?
07:52.55billatqbenjk: Probably free
07:53.08benjkcare to let us host it ?
07:53.14trixterbillatq: I made a glass of cocoa to drink
07:53.15trixterit was tasty
07:53.22benjkon the Mac Asterisk community site?
07:53.27zobiabenjk.
07:53.36benjkhttp://www.astmasters.net
07:53.49trixterjust dont say the domain name too quickly
07:53.50trixter:P
07:53.56billatqWell, I've already got a nice place to host it
07:54.00benjk:-D
07:54.00billatqAnd it's not quite done yet
07:54.08*** part/#asterisk SERGEUS|WORK (n=SERGEUS@ippe-245.ippe.ru)
07:54.52benjkwe could bundle it with the Mac Asterisk installers I release
07:54.52billatqhmm
07:54.52billatqthat might be cool
07:54.52benjkthis way it would be on every Mac that runs Asterisk
07:54.54benjkby default
07:54.54billatqHehe
07:55.06billatqYeah, I've been annoyed at the lack of a decent iax client for mac os x
07:55.06trixterbill: do it you will be famous!
07:55.10benjkwell, almost, cause some folks build themselves
07:55.21implicitbenjk, do macs that run asterisk on linuxppc count?
07:55.22trixteroh wait do we want another bill to be famous for computer software?  the last one didnt turn out that well
07:55.34billatqtrixter: Heh, I worked for that bill this summer
07:55.35benjkChristian Draghici of Romania has released a nice Cocoa IAX client
07:55.43billatqOh?
07:55.45benjkLoudhush
07:55.55benjkbut it is shareware 16.99 USD
07:56.01trixterimplicit: no I dont think so cause linuxppc doesnt do cocoa :P
07:56.04ComputerWarmok next problem.... does anyone know why i would be getting this error  record_exec: No extension found ?
07:56.08billatqYeah, and it's not really that pretty
07:56.17implicittrixter, :)
07:56.28benjkLinixPPC runs Asterisk fine
07:56.30implicittrixter, do you like GTK2?
07:56.36implicitbenjk, but not cocoa fine
07:56.37implicit:)
07:56.42benjkeven with hardware support, zaptel etc
07:56.49implicitbenjk, i know
07:56.55benjkbut you can run OpenStep
07:57.12benjkand port your Cocoa apps over fairly easily
07:57.17billatqYeah
07:57.19billatqActually..
07:57.22implicit*your* cocoa apps
07:57.23billatqfor this app, that should be easy
07:57.39implicitgoodnight benjk
07:57.39trixterimplicit: I nbever got into gui stuff that much..  I wrote 2 windows programs that were gui based and a few that werent..  The two that were one was 1994 and one was 2005.  I dont do it often..  in unix I have written 3 or 4 gui programs, mostly 94-96..  so I cant comment on gtk2
07:57.50trixterif it doesnt run in an xterm what good is it? :P
07:57.55benjkgoodnight?
07:57.59implicittrixter, exactly my thoughts
07:58.03implicitbenjk, yes i'm tired
07:58.07implicitbenjk, going to sleep
07:58.08implicitsoon
07:58.23billatqSo all I need to do is implement incoming calls (maybe a few hours at the point I've reached now)
07:58.31implicittrixter, if you like mysql5 say so
07:58.34billatqhold and caller id are pretty much ready
07:58.37benjkok, fair enough, I thought it was something I said ;-)
07:58.40trixteroh I have used java to do gui stuff including games, forgot about all of that
07:58.46implicitbenjk, not at all :)
07:58.58benjkbillatq: you should also implement Bonjour (aka Zeroconf)
07:59.08billatqbenjk: How'd that work?
07:59.12billatqAsterisk advertising it?
07:59.27trixterimplicit: I think mysql 5 has some really nice and long awaited features..  better subselect support, stored procedures !!! that is a big one that has been missing
07:59.33benjkwe made a module for Asterisk to advertise SIP and IAX services over Bonjour/Zeroconf
07:59.37billatqoh
07:59.38implicittrixter, i think it is very nice
07:59.38billatqsweet
07:59.45implicittrixter, you forgot views too
07:59.57benjkhttp://www.astmasters.net/projects.html#zeroconf
08:00.01billatq(Do any clients support that?)
08:00.01implicittriggers, lots of cool shit
08:00.14benjkyes Loudhush does I think
08:00.33implicittrixter, see you tomorrow
08:00.39benjkand the SFLphone guys are working on support as well
08:00.51billatqnifty
08:00.53*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:01.18benjkand I am trying to get a hardphone manufacturer to support it too
08:01.28benjkApple would love to see that
08:01.39billatqYeah, wouldn't surprise me
08:01.44benjkthey said something about talking to Cisco about it
08:01.59benjkthey want a Zeroconf supporting WiFi phone
08:03.11benjkbillatq, are you on our mailing listyet?
08:03.35*** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net)
08:03.40billatqbenjk: Nope, that I am not
08:03.52billatqI just started this project today and am surprised I already have something working
08:04.00benjkMac Asterisk Mailing List
08:04.24bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
08:04.51*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
08:04.56benjkyeah, tainted was asking for a Polycom phone to bvuy today
08:05.23escribzzheh polycoms are a pain :)
08:05.42oriontknhas anyone used the Grandstream GXP-2000
08:05.42oriontkn?
08:05.49benjkhey, I don't care, I just said somebody was asking for it
08:06.12billatqI like my $30 SIP ATA
08:06.19benjkorion: my associate in South America has ordered tow of those for a demo next week
08:06.21billatqEven has two FXS's
08:06.50oriontknI have more SIP ATA's than I know what to do with
08:06.56oriontknI think its 2 per sq foot now
08:07.10benjkbillatq: you should sign up to our mailing list http://www.astmasters.net/maml.html
08:07.17trixterI would love to have a $30 ATA with 1 fxo and 1 fxs but no one makes one
08:07.34oriontknI unlocked the Sun Rocket Gizmo.. and have some of those 2 linue AC-211's laying around...
08:07.46oriontknjust would like to unlock my Linksys RT31P2's now
08:08.07benjkNTT gives out SIP ATAs for 300 yen or so I am told
08:08.15benjkthats about 3 USD
08:08.19billatqbenjk: Yeah, just did
08:08.19oriontknomg
08:08.39benjknot sure if they are locked though
08:08.42trixterum what brand ata?
08:08.47benjknever used one
08:08.47billatqBest Buy right now has a deal where you can get the AT&T callvantage phones, then reflash them with the sip firmware
08:08.50benjkNTT
08:08.51trixterfor $3 I would invest some time to unlock it
08:08.57trixterI am doing that with my mot vt1000 locked to vonage
08:09.01benjkNTT make their own equipment
08:09.09benjkmost of it
08:09.09billatqSo I've got two D-Link ones that I bought for $30 each
08:09.22billatqWhich work great with asterisk, though it took a while to get the settings just right
08:09.30trixterjust so that doesnt go the way of the vonage pap2s
08:09.34oriontknthey are dlink ata's?
08:09.48trixterwhere you could get a pap2 for like $20-30 for vonage, and for a while it was trivial to unlock
08:10.04trixterthen they changed it and you cant as easily unlock em ...
08:10.12billatqhttp://www.anatifero.us/weblog/2005/11/09/linksys-dvg-1120s/
08:10.15billatqThat's my blog post on it
08:10.39trixterI will talk to my friend about that he wants a hardphone of some type
08:10.42zobiahello benjk
08:11.22trixteraggie?
08:11.25trixterahem
08:11.30[chico]Hallo spricht irgendwer deutsch??
08:11.35trixterI remember quite a few stories I have heard about aggies
08:11.45billatqHeh, there do seem to be a number
08:11.52lme[chico]: better in english
08:12.04trixtercourse I actually know where college station is so I am more qualified to speak on aggies :P
08:12.11*** part/#asterisk camonz (n=camonz@200.8.21.123)
08:12.27benjkschwiizertuetsch
08:12.32*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)
08:12.33billatqtrixter: You'd probably have more to hold against me for working at Microsoft than for being an aggie, I'd think
08:12.35trixterI lived in texas for like 7 years :)
08:13.04billatqThe place is kind of scary, the stuff just kind of works
08:13.21trixterMS or college station?
08:13.42billatqMS
08:13.45trixterahh
08:13.58billatqMaybe that's why a lot of their stuff sucks--they don't realize how bad it is in the Real World
08:14.29trixterwell I am friends with 3 people at MS..  a tech writer, a project manager and some random person that flies around the world doing odd stuff..  I think he is somehow in support but I really dont know he doesnt disclose his title I just know what he used to do
08:15.11billatqHaha
08:15.29oriontkntrixter: do you work for VA Linux now?
08:15.53billatq(I'm actually hiding out at Hewlett-packard as an intern for the time being right now)
08:16.13billatqSo I get to not be evil at the moment
08:16.38benjkbillatq: is that why they got rid of Carly?
08:16.43benjk:-)
08:16.58billatqbenjk: Heh, they got rid of Carly because she was crazy
08:17.11billatqRumor is that they're giving everyone nice bonuses this year
08:17.12trixteroriontkn: why do you ask that?
08:17.22benjkYeah, that's about right
08:18.09oriontkncuz I knew of someone that lived in texas and worked for MS and I think moved on to VA....
08:18.14oriontknsometimes its a small world
08:19.08oriontkngrr...   I'm still getting 'Registration Refused' trying to IAX to FWD from *
08:20.30trixterI lived in texas from about 1980-1987 or so
08:20.45trixterodds are you didnt know me
08:20.45trixter:)
08:20.52oriontknyea...
08:22.11trixterhrm it had to have been only 6 years I was there
08:22.22benjkyeah I didn't know you either, what a coincidence
08:22.26trixterjust counted the years of school I had during that time
08:22.40trixterbenjk: yeah total coincidence!
08:23.56benjk:-D
08:24.05benjkHey guys, this is weird
08:24.47benjkthis German chap has no DNS (which is why he cant see his SIP provider) but he managed to get on to IRC
08:24.50trixterdoes anyone know if ethernet.org has produced any java based games?  or released any games of any kind, particularly 20 year old games rereleased?
08:24.53trixterlike pacman and stuff
08:24.59benjkhow would he have done that?
08:25.09trixterIP
08:25.26trixtermaybe the dns issue isnt 100% maybe one got through
08:25.36trixtermaybe he has something in /etc/hosts
08:25.43benjkdoesn't look like it
08:25.54benjkweird
08:27.28trixterI gotta verify that the bluetooth headsets are still $20-30 from this one place
08:27.33trixterthey were on sale and I dont know if its over
08:28.54oriontknwhat in /etc/resolv.conf
08:28.58oriontknwhats in?
08:29.22benjkyeah I asked him, he needs to find it I guess
08:32.41benjkquite amazing how far some people get with Linux even if they haven't got the faintest idea about anything at all none whatsoever, got to give him credit for that
08:33.08oriontkntry that in the days of kernel 0.98
08:33.22benjkthat takes stamina ;-)
08:33.56trixteralmost got the 500,000 routes done ...  listing geographic, mobile, special services, short codes (911, 999, etc), sure beats the old list of about 5500 :)
08:34.18trixterI should be done with that tonight for anyone that wants a copy ...
08:34.28oriontknsign me up
08:34.36P4C0I have been using sjphone soft sip phone, but it's not open source and I have publicity each time I get a call, otherwise it's really cool, does anyone knows a good replacement (open source or at least without publicity and with more or less the same features)?
08:35.06oriontknI'm trying IdeFisk with IAX
08:35.10oriontknI just downloaded it
08:35.19benjktrixter: well done!
08:35.29benjkwhats IdeFisk?
08:35.34P4C0for SIP?
08:35.45P4C0benjk: a software phone
08:35.46oriontknWin32 IAX softphone
08:35.58oriontknhttp://www.asteriskguru.com/tools/idefisk_beta.php
08:35.59P4C0oriontkn: why IAX?
08:36.00benjkpublicity?
08:36.08benjkAh
08:36.11oriontkn>heavy sigh<
08:36.15benjkIAX rocks!
08:36.24P4C0better than sip?
08:36.31oriontknI'm trying to get an Asterisk box to talk to Free World Dialup
08:36.31benjkIAX totally rocks!
08:36.38oriontknand it keeps rejecting my registration
08:36.48trixterbenjk: the worst thing was the fact that the server wont let me download everything I have to get html and parse it.  the lkongest part of this is their stupidly written webapp that goes slower as the page count gets higher
08:36.53P4C0benjk: why?
08:36.53oriontknso I wanted to try connecting from another source... to independantly verify the problem
08:36.55oriontknwhich I did
08:36.58benjkSIP isn't really all that good you know
08:37.04benjkit's overhyped
08:37.19benjkSIP is 2nd generation VOIP
08:37.30benjkIAX is 3r generation VOIP
08:37.36benjker 3rd
08:37.37*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
08:37.44P4C0what can I do with IAX that I can't with sip? (only one... well maybe 2 :)
08:37.56billatqEasy setup, especially using NAT
08:37.59benjktraverse any number of NATs
08:38.00oriontknnat without have to forward ports
08:38.08oriontknno STUN
08:38.10{zombie}with IAX you can trunk calls - saves bandwidth
08:38.15trixterI dont use stun with sip and it works
08:38.20trixterthrough nat
08:38.24benjksurvive the nastiest internet environment ever known to mankind
08:38.28*** join/#asterisk chapeaurouge (n=chap@85.201.80.249)
08:38.44chapeaurougehmm... the xchat auto-connect joins the channel b4 registering my name... this sux..
08:38.47chapeaurougehi all
08:38.52trixtertrunking calls only saves bandwidth in special circumstances, it doesnt save that much either..  and can actually slow stuff down under some circumstances
08:38.53P4C0wait, I may be wrong,  but the problem with nat is not sip is the rtp and rtcp packages... right?
08:38.56trixterslow all calls ratherthan just one
08:39.01benjklike when your traceroute changes by the second and you never get the same output from it twice
08:39.25benjkPC40 the problem with SSIP ios that it isnt actually a VOIP protocol
08:39.39benjkit only deals in introductions
08:39.43*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:39.53benjkthen your clients are on their own
08:40.03P4C0benjk: yes I know, it only does the signaling and the rtp and rtcp carries the voice packages, is that the same with IAX?
08:40.33trixterdoes iax support CRTP?
08:41.00mog_homecrtp?
08:41.04mog_homeiax doesnt do rtp
08:41.09benjkno IAX is a real IP protocol
08:41.20trixtercause a normal RTP packet is like 40 bytes, CRTP is 2 bytes, much better..  if iax doesnt do CRTP then you would have to trunk 20 calls together to break even on the bandwidth
08:41.28trixterAhh I see
08:41.34benjkit separates signaling and payload by envelop not by using different ports
08:41.59trixterby saying iax is a real IP protocol you mean at the udp layer or is it a subtype to udp?
08:42.03benjkSIP is a relic of the circuit switched world's way of routing phone calls
08:42.38benjkby real IP protocol I mean that it is a protocol that doesn't break the internet paradigm\
08:42.44benjkTCP/IP paradigm
08:42.54trixterwhat is that?
08:43.00P4C0humm well my provider will give me SIP (althrou I'm almost sure that they are using asterisk) I can ask them to see if they support IAX, cause there's no point in having my local phone with AIX... I mean they will be inside my local network... or am I wrong?
08:43.06trixterplease explain becuase um I didnt realize that sip did
08:43.08benjkwhich is that you can route any packet by oinly locking at the envelope
08:43.26trixteryeah that is how RTP works
08:43.30benjkSIP/RTP break that paradigm
08:43.37trixterhow so specifically?
08:43.44benjkit has to do packet inspection
08:43.50benjkin verisou cases
08:44.02trixterthe packet is routed by the IP header to the destination machine if that didnt happen EVERY router on hte net would have to be RTP aware and they arent
08:44.20benjkit is akin to the post office having to open your letters to know how to delivere them
08:44.27h3xbenjk: So you are saying that FTP isn't a real IP protocol?
08:44.31P4C0trixter: because sip use some ports (5060 - 5070) for signaling, placing call, invites rings and stuff, then it just tell rtp and rtcp to carry the voice like a totally different stuff
08:44.34benjkthat'; precisely the problem
08:44.39trixterplease name a situation where packet inspection of more than the IP header is required, short of final delivery where the udp packet has to be looked at
08:44.53Nixbenjk: thats why SIP scales..
08:45.09Nixactually thats why RTP based VoIP protocols scale and IAX doesnt ;-)
08:45.10trixterno you dont have to look at more than the IP header to know which machine to route a sip packet to
08:45.23benjkthe separation into different data streams that the router is unaware of is breaking the parading of the internat as a stupdi network
08:45.32h3xThe point of having RTP seperate is so that your media stream can be direct between two devices
08:45.33trixterI would like you to state 1 situation where more than the IP header is required short of final delviery where the udp packet has to be looked at and stuff..
08:45.36*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
08:45.38h3xand have proxies and what not in the middle
08:45.41mog_homeugh iax scales, and in one of the more recent rfxs there is a plan to allow for cloning of the signalling
08:45.47benjkIAX eliminates nodes in between within 8-10 seconds
08:45.48mog_homeerr rfcs
08:45.49trixterso you are saying that ftp breaks the paradigm
08:45.51trixterno it doesnt
08:45.59trixterand hasnt for the 20+ years the ftp protocol has been a standard
08:45.59h3xThe only reason IAX is integrated is to deal with NAT better
08:46.14h3xyeah
08:46.14benjkIAX has been submitted to the IETF or so Frank Miller told me this week
08:46.23h3xyou can have a FTP client connect to two different FTP servers
08:46.29h3xand it can tell one server to upload to the other
08:46.33h3xusing the PROXY commands
08:46.42benjkbesides there are other problems with SIP
08:46.48h3xof course, most people dont know about this, besides warezers... heh
08:46.50trixternormally it isnt the proxy command that isnt a standard one
08:46.55benjkMake a call via SIP
08:46.58trixterthe protocol itself gives IP and port to connect to
08:47.06P4C0actually when I find out that sip works like that my first question was... why!?
08:47.07benjkput your party on hold
08:47.10h3xtheres more problems with IAX than SIP
08:47.21benjkthen disconnect your sip phone
08:47.21mog_homeyou havent used iax then h3x
08:47.27benjkand reconnect
08:47.28trixterhowever I stil lhavent heard anyone give a single example of how sip breaks the IP paradigm, I have heard peopel say htati t does but nothing to substantiate that, I would really like someone to give just one example
08:47.38h3xhah ive been using asterisk for 2.5 years
08:47.43trixterwhile you think about it I will go smoke, be back in 5
08:47.45benjkthye call will be on hold forever
08:47.45h3xIAX dosent have t.38 or anything like it
08:47.53mog_homeits not meant too
08:47.55P4C0trixter: I did
08:47.56trixtert.38 is a codec like g.729
08:47.57h3xIAX dosent have a way to seperate media streams
08:47.59mog_homesip can do im, tv, etc
08:48.04trixterp4c0 please repeat it then
08:48.06h3xyou can run SIP through STUN and get one stream
08:48.08mog_homewe didnt want that with iax
08:48.11benjkSIP has no f***ing way of knowing about it
08:48.12mog_homeiax is for voice
08:48.19h3xIAX dosent have a video specification
08:48.27trixtert.38 is also very bandwidth intensive becuase it sends this and last packets data..  t.37 is a better way to deal with faxes
08:48.30benjkfax will die anyway
08:48.36h3xIAX dosent interop with anything but asterisk and asterisk soft phones
08:48.42benjkwho needs fax in a wired world
08:48.46mog_homefax wont die as long as email isnt legally binding
08:48.47P4C0trixter: sip use ports for signaling and different ports for rtp, and it's not like ftp
08:48.53benjknonsense
08:48.58trixteruntil fax dies and it wont die for YEARS t.37 is a better way to go les bandwidth
08:49.05benjkthere are quite a few IAX servers
08:49.05mog_homeh3x you have obviously formed a strong opinion on this
08:49.13mog_homei am no mood to go zealot war with you
08:49.16trixterAhh so ftp uses 2 ports, one for signaling and one for data
08:49.20benjkone for Solaris that runs on a 64 CPU box
08:49.21trixterjust like you said sip does
08:49.22trixterI see
08:49.25mog_homebut iax2 is really well designed
08:49.29benjkwritten in Java
08:49.29h3xmog_home: I can't see sending 25,000 calls through an asterisk box
08:49.31trixterhow are they not the same again?
08:49.34h3xor a cluster of asterisk boxes for that matter
08:49.37mog_homeand alot of your complaints are going to hapen
08:49.42h3xit'll work with sip though
08:49.46h3xif you offload media
08:49.58trixterlooks to me like sip is doing stuff the old school internet way
08:50.01benjkIAX and LTP are the next generation VOIP protocols
08:50.06*** join/#asterisk roulduke (i=raha6ktk@p508D2AF8.dip0.t-ipconnect.de)
08:50.10P4C0trixter: yes, I got your point
08:50.12trixtersip also supports host redirection like ftp does
08:50.17h3xyes
08:50.21mog_homeh3x there will be soon a way to clone a signalling stream and then do a native transfer
08:50.27Nixbenjk: Is your name "slashdot troll" by any chance? :-)
08:50.29trixterI dont see how doing stuff the way its been done for 20-30 years is breaking any paradigm
08:50.29h3xasterisk has better performance with SIP than IAX2
08:50.31mog_homethus getting you the 25,000 scale
08:50.34benjkseparating signaling and payload over different ports is sooooo 1960s
08:50.50mog_homeand i disagree, if you arent doing billing
08:50.56Nixbenjk: using NAT is sooooo 1960s!
08:50.59trixterum in the 60s didnt they do inband signalliung?
08:51.04h3xthe only reason you have 25,000 ports is if youa re billing somebody :P
08:51.04mog_homeits sooo much easier to allow asterisk boxes to do native bridging
08:51.09h3xunless you are skype
08:51.15mog_homeumm no, what if you are ibm
08:51.16mog_homeetc
08:51.19benjkthe point is that the internet is a stupid network . perios
08:51.20mog_homeinteroffice calls
08:51.33h3xwell heres the problem
08:51.35benjkthe PSTN is an intelligent network
08:51.40h3xasterisk is a shitty high capacity gateway
08:51.41P4C0benjk: the more stupid the best!
08:51.52h3xyou cant turn a DS3 into IAX2
08:51.54Nixbenjk: you still didnt explain why SIP and the internet being stupid have any relationship
08:51.55mog_homeasterisk != proxy gateway
08:51.58mog_homei think you can
08:52.01benjkSIP is designed based on the PSTN concept
08:52.08mog_homehow many channels is ds3 600 or so right
08:52.10benjkI am explaining it right now
08:52.15trixter672
08:52.19Nixits actually "smart" routers like NAT devices that cause problems with sip.. not stupid ones
08:52.21trixterin america anyway 28 T1s
08:52.25h3x672
08:52.36benjkstupid network doesn;'t mean its a bad network or anything like that
08:52.37mog_homei think you can do it with a beefy box
08:52.37P4C0benjk: yes, but that dosen't help the nat problems...
08:52.40trixterI think the DS0 count is the same in europe but a lower quantity of E1s
08:52.43h3xnot with compression
08:52.46mog_homei saw a box do 1000 calls
08:52.52h3xand licenses for g.729 would be $6720
08:52.55mog_homesip, and monitoring the calls so it was doing the media
08:53.03Nixbah.. time for work
08:53.05*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:53.14mog_homewell you arent gonna do that many g729 currently
08:53.15h3xit costs about $12k to buy a ds3 MAX TNT DS3-> Ethernet
08:53.18mog_homeat least on a box
08:53.20benjkit means that the paradigm is that the intelligence is at the edge of the network, not built into the network
08:53.35h3xsimilar economics on any other large scale system
08:53.37trixterh3x: yeah..  although if you are pushing 672 calls (on a DS3 you would be able to push way more than 672 if you did g.729) you should be able to afford the $6k
08:53.41benjkSIP needs intellignece built into the network
08:53.46benjkIAX doesnt
08:53.58benjkthats how SIP breaks the paradigm
08:54.08trixterSIP need less intelligence
08:54.10h3xnow that is some /. blabber
08:54.14P4C0benjk: that's true
08:54.21trixterits the stuff that is trying to be more clever instead of one ip one machine the way it used to be
08:54.23trixterthat is what breaks it
08:54.35benjkSIP needs an intelligent network infrastructute
08:54.35trixterthe more clever stuff is the more likely stuff is to break
08:54.42trixterno it needs a very very stupid one
08:54.57h3xtheres not a single good IAX hard phone yet
08:55.01mog_homeyeah
08:55.02trixterthe more intelligent it is the more it tries to be creative with packet delivery instead of just tossing packets based on ip
08:55.07benjkit cannot work in a network where intelligence is only at the edges
08:55.08mog_homeits only a few years old h3x
08:55.15mog_homeand doesnt have a standard yet
08:55.24mog_homethat usually means people dont make hard phones with it
08:55.24h3xonly one of the IAX2 soft phones (diax) supports SendURL and some other IAX proprietary features
08:55.34h3xwell, gnophone too but it sucks
08:55.40benjkIAX works well in a network where all the intelligence is at the edges
08:55.51benjkand none in the network
08:56.16h3xbut the point is, in north america at least, all our big ass carriers have deployed voip
08:56.18benjkread up on the stupid network stuff at David Isenberg's website
08:56.19h3xwhy buy gateways
08:56.22trixterthat is more intelligence than sip needs or wants, unless you are just talking about bgp or something at the edges, sip doesnt care about that
08:56.26h3xthey already have SIP stuff
08:56.36mog_homethey already had h323 stuff too
08:56.40mog_homebut there is this thing
08:56.42mog_homecalled progress
08:56.45mog_homeit happens
08:56.45benjkbgp is not at the edge
08:56.57benjkbgp is inside the network
08:56.57h3xbut they spent millions of $
08:57.00trixterborder gateway protocol isnt at the borders or edges?
08:57.03h3xbuying them sonus boxes
08:57.04mog_homei agree
08:57.07benjkthe edge is the UA
08:57.10mog_homebut over time you upgrade
08:57.22h3xto what?
08:57.22P4C0well let's put one example, a problem that I have, one isp in miami, it give us a private ip address and we had a pool of public ips to access internet... (random)... it was really difficult to make a call to that phone... almost imposible, the only way was to make the call withing 30 minutes from registartion
08:57.22mog_homejust like there are still tons of h323 gateways out there
08:57.25trixterI dont know what you mean by edge then if something that deals with the handoff from one network to another, thus is at the very edge of a network doesnt count
08:57.28h3xyou cant replace sonus with asterisk
08:57.28mog_homebut no one buys them anymore
08:57.35benjkthats a deliberatly chosen misnomer
08:58.05benjkthe edge is the end of the communications chain
08:58.12benjkthe end point
08:58.24benjka bgp is in between the end points
08:58.36P4C0now, if iax can handdle that without failing like sip, I think its better
08:58.38trixterp4c0: ok, so sip doesnt like people being clever and trying to overengineer the problem..  and I can understand that iax works well on kids networks where sip works better on grownups networks..  to be honest an isp that allocates rfc1918 addrs is stupid
08:58.41h3xlarge carriers arent gonna be deploying iax2 any time soon
08:58.42trixterjust plain stupid
08:58.44benjkand therefore it is per definitionem not the edge
08:58.51trixterthey cant want to stay in business
08:58.58trixtereven aol stopped doing that 10 years ago
08:59.32trixterbgp is from one router that is the last network device on one network and to a router that is the first on a new network
08:59.33trixterit is the border
08:59.43trixterthat is not a misnomer that is how its deployed all over hte globe
08:59.45h3xit does matter whats on the edge
09:00.00benjkit is not the endpoint of the communications stream
09:00.01trixterwell he said that bgp didnt count as the edge I didnt know what did then
09:00.03h3xwhen you are going from somethign thats RTP to something else that dosent have RTP
09:00.11trixterthe fiber wire that connects the two routers?
09:00.11benjkthe SIP phone is the end point
09:00.27trixterahh ok I see now what you are saying
09:00.29P4C0trixter: yes but some isp do that and you need to find a way to work around
09:00.40h3xThe point I am making is its silly to get TDM handoff from a major carrier
09:00.40trixterdifference between edge of the network and end of the communications stream
09:01.08h3xbecause today, right now, you can order a dedicated IP circuit with SIP on it
09:01.08trixterso what is the header size of iax2
09:01.13trixternot counting codec payload
09:01.21trixtersip can work with 2 byte header sizes
09:01.24h3xwell i do like iax2 trunking
09:01.31trixterthat includes IP, UDP and RTP
09:01.43benjktrixter that doesn;t matter much when we are talking about concepts
09:01.45trixterregular is 40 bytes, CRTP is 2 bytes ip down
09:01.57mog_hometrixter go to asteriskgurus site
09:02.01benjktake LTP as another example of an IAX like protocol
09:02.03h3xapparently you can jam 180 channels with G.729 into the space of a single data t1
09:02.04P4C0which of the two uses less bandwith??
09:02.05mog_homethey have a caclulator
09:02.13mog_homeiax2 trunked is 1/3 bandwith
09:02.15mog_homeat 100 calls
09:02.17benjkit is more llightweight than anything else
09:02.21trixteriax2 trunking doesnt look so good from a provider point of view because it only works on multiple calls from a given endpoint, and most calls re from different points
09:02.47h3xtrixter: Thank you, thats where im going with this
09:02.49trixtermog_home: I have been talking with the askguru site to get them to upgrade their bandwidth calculator to make it more accurate
09:02.56mog_home?
09:02.57trixterso refering me there isnt that good of an idea
09:02.59mog_homewhats wrong with it
09:03.12mog_homeive never gotten an inacurate number from it
09:03.17trixterit doesnt count a lot of stuff that should be
09:03.30trixterI bet you have and you didnt notice it
09:03.30mog_homelike
09:03.49trixterlet me go smoke, but if you use dsl, t1, t3, oc1, oc3, etc you got inaccurate info from it
09:03.55P4C0I think I'll use sip for now ;)
09:03.59trixterI will explian or you can look for my posts to the mailing lists lately about that
09:04.09benjkwell you better not use that bandwidth as an argument for SIP because iof you take LTP into the equation you will be veru disappointed
09:04.24trixterno sip has the same problem
09:04.31benjkand LTP is very similar to IAX
09:04.34h3xWhat needs to happen is adding some stuff to sip
09:04.51P4C0but, the problem is that I need a good opensource sip phone... for windows and linux, and so far I haven't found anyone...
09:04.54h3xand autonegotiate options
09:04.56trixtergimmie 5 minutes and I will explain..  I have been saying I would smoke for like 10 minutes now :P  be back in 5
09:05.09konfuzedoh debian asterisk/unstable is very nice
09:05.13benjkso its the concept of not separating signaling and pyload as if it was a circuit switched network which counts
09:05.41benjkSIP is basically a design done by circuit switched thinking engineers
09:05.59benjkoh we need a signaling channel, that's uhm, port 5060
09:06.34benjkoh we have 10.000 trunks coming in here, thats uh, ports 10.000 and counting up
09:06.48P4C0when someone calls my sip phone how is the one that send the first rtp or rtcp package?
09:07.15benjkthats how you would design a protocol if you are breathing circuit switched, if you have never seen TCP/IP before
09:07.37P4C0benjk: 5060 to 5070 I think... and for rtp 10 000 to 25 000 (tell me if I'm wrong)
09:08.07benjkgood TCP/IP design means that the application layer takes care of unbundling the data and interpret what it means
09:08.18benjknot the port numbers
09:08.28oriontknthen you use 1 port number
09:08.29benjkthats so silly
09:08.37oriontkn;)
09:08.42benjkthats; what IAX and LTP do
09:08.51benjkand SIP cannot
09:08.52P4C0benjk: sorry I'm thinking about nat, firewall and all that nasty things
09:09.15benjkthis is also of consequence for the NAT traversal wscenarios
09:09.33benjkif you have only a single port, then NAT will woirk
09:09.47benjkso its all coming together
09:10.03P4C0benjk: totally agree, even if you have two port it will work... but with random ports... it's hard
09:10.11benjkIAX woerks well (incl NAT traversal) not because its magic
09:10.18h3xSIP can do it with STUN fools!
09:10.20trixterback..  ok the reason that asteriskguru is wrong is that it doesnt include things like atm framing
09:10.27h3xheh
09:10.34benjkbut because it is designed following the TCP/IP philosophy
09:10.42trixteratm uses a 53 byte fixedcell size, therei s a 5 byteheader and an 8 byte sar trailer
09:10.52trixterthose have to beincluded pppoe uses an additional 6 bytes as well
09:10.59benjkapplication layer does that interpretation of whats in the packets
09:11.07trixterany packet sent that doesnt evenly fall in is padded so extra bandwidth is used
09:11.12benjkand envelopes say whats in the re
09:11.32benjkthats why it is called a layered protocol
09:11.43trixterso g.726 with 10ms samples is 40 bytes plus the 40 byete IP UDP RTP header..  that means 80 byes need to be sent and results in 8 bytes of padding per packet
09:11.56benjkno need to use different ports for a connection
09:12.01trixterwith pppoe its only 2 bytes but different codecs can have different amounts of padding, wasted bandwidth
09:12.34P4C0trixter: is there a protocol that doesn't need padding??
09:12.56trixterif you havea crtp stream plus 9.6 ms with g.276 you will fill evenly 1 atm cell (no pppoe)
09:13.02P4C0benjk: well, maybe there is a reason why
09:13.12trixterone other thing that asteriskguru doesnt include is variable sample sizes
09:13.24trixterwhich if you sample too quickly you have more overhead so the ratio is worse
09:13.34trixterand that doesnt include any IP header options which may exist but generally shouldnt
09:13.42P4C0trixter: did you notify that to asteriskgurus?
09:13.43trixterp4c0 I said before I left that sip has the same problem
09:13.53*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:13.58trixteryes via a long email exchange
09:14.08trixterhowever its REALLY hard to compute that becuase of all the variables
09:14.08P4C0trixter: :)
09:14.19trixterand it was to be sent off to the coder who is responsible to that to update it
09:14.25benjkahahaha
09:14.36benjkthat's the statemwnt of the day
09:14.44benjkits gotta be
09:15.08trixterin effect you have to add like 10 check boxes for the different options to get an accurate count of how much raw bandwidth is used per channel so you can see how many channels you can support without overlap, take an oc3, you think you have 155Mbps but if there is a lot of padding that can go down
09:15.08benjkPRICELESS!
09:15.46P4C0so there's no use for that... I mean if you waste a couple of bytes is not that important... trying to optimize to the point of no bytes lost and no useless or not needed bytes is pointless and can't prevent scalability, don't you think?
09:16.01trixterin effect on an atm network a g.726 10ms packet is only 76% efficient
09:16.19ComputerWarmquestion can asterisk play live streams for music on hold?
09:16.22mog_homeexactly, atm sucks
09:16.23trixterturning your oc3 into effectively a 117Mbps link
09:16.28mog_homeyes
09:16.37trixtercomputerworm: yes I do it all the time
09:16.46trixterand with ices it can create streams too
09:16.51trixteralthough ont from music on hold
09:17.15ComputerWarmtrixter ok could you show me your line please?
09:17.53trixteryou either have to create a custom moh line or touch a file that is a url in your moh directory
09:18.08P4C0talking about music... is there a soft phone that put on pause my xmms, amarok or winamp will I talk?? :p
09:18.08benjkhey no indecent proposals in here please
09:18.13trixterthat way when mpg123 gets the file list it will see 'http://...'
09:18.20trixterif you pass a url to mpg123 it will stream it
09:18.43ComputerWarmoh ok great thanks
09:18.57trixterp4c0: xmms -t will pause/resume xmms, if you cna get it to execute that command on answer it will work, I dont kjnow of any that will automagically execute on answer
09:19.10trixterComputerWarm: there are examples on voip-info.org about that specifically
09:19.27ComputerWarmok thanks i will look there for more info. if i run in to a problem
09:19.30P4C0i will be nice to have a death metal on hold :p
09:20.05P4C0then it will make sense to put people on hold :D
09:20.24*** join/#asterisk zoa (n=kkk@213.16.46.130)
09:20.26trixterheh
09:20.28zoahey ho
09:20.32trixterstream the hampster dance instead
09:20.51benjkdeath metal?
09:21.34trixterkeep in mind that streaming copyrighted material for hold may be illegal, you may be required to pay royalties for any music that you put on hold music..  bmg is like $200/year for their content, not that bad, but there is a ton of royalty free music too
09:21.54P4C0can I have like different kinds of holds music??? a normal hold for important people, and a really anoying hold for the rest?
09:22.06trixteryes
09:22.12benjkthe best is you use classical music
09:22.28benjkthat's difficult to match to any particular label
09:22.29_Lyfe_or, a normal hold for most people, an annoying hold for people you hate, and a really cool hold for important people :)
09:22.37trixteryou can create different classes and then all you have to do is set the correct class per user
09:22.44P4C0benjk: naaa give me a mic and I can rec a hold music that will really make you go crazy after 5 seconds
09:23.02P4C0_Lyfe_: that will be even better
09:23.08benjkNo thanks I'll have a Guinness instead
09:23.24*** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au)
09:23.45P4C0:)
09:23.48_Lyfe_or, as in my scenario, when i call the guy i work with, and he puts me on hold (which is way to frequent), then i'll have to have it play something othe rthan the default hold music we have which drives me insane.
09:24.21trixterit would be better if you set it so that the user can select their hold music
09:24.35P4C0_Lyfe_: yes, hold music is important, I mean how many hours a day you are on hold??
09:24.49benjk10 hours or so?
09:24.51oriontknan IVR to manage your MOH
09:24.55trixterwhich is possible if you set it up to transfer someone then transfer em back when you wanna talk, and transfer em to a context that lets them select either via mp3playback (which suports urls) or something
09:25.08_Lyfe_P4C0: dunno.. i probably spent about 30 minutes on average durin a conversation on hold with that guy, cause i call him up, and he's bouncing off to answer a call, or whatnot.
09:25.17ComputerWarmtrixter question it says it start music on hold but its not playing the music?
09:25.27benjkHey, what are you doing on the phone all day long
09:25.45benjkMy iPod is away for repair, I am listen to the music
09:25.48_Lyfe_attemping to find out what i'mm supposed to do next ;)
09:25.56P4C0hehe
09:26.17oriontknuse a sip phone with local hold...
09:26.17P4C0a good one can be "you are on hold... you are on hold" over and over again ;)
09:26.20benjkCan I borrow anouther phone plase?
09:26.24benjkWhy?
09:26.36benjkBecause I like to listen in stereo
09:26.39_Lyfe_or, you can just play dueling banjos (that's enough to drive most people nuts) :P
09:26.45oriontknwhen he puts you on hold have a packet come back that says you've been placed on hold and have it unpause and pause your winamp
09:26.52P4C0benjk: get a good headset :p
09:26.55trixterp4c0: with that anoying french guy that changes his inflection greatly
09:27.06trixterthe one that goes 'YEEEEEEEEEEEEEEEEEEEEEEEEEEEEES' and stuff
09:27.50P4C0trixter: hahahahahahaha sure, mmmm I can do that... if my boss say something I will blame the hackers ;)
09:28.38trixteryou know who I am talking about right?
09:28.39_Lyfe_make your hold music play random "phone errors" .. "I'm sorry, your call cannot be completed as dialed" )
09:28.45trixterthat guy even made it into the simpsons in one episode
09:28.56P4C0trixter: yes I think so
09:29.01trixter"weasels have eaten our phone system"
09:29.17P4C0erotic hold?
09:29.21trixterp4c0: well make him say 'yes you are on hold' in htat voice
09:29.24trixterthat would be quite anoying
09:29.41P4C0trixter: yeah, the trick is the "over and over again" stuff
09:29.52trixterwith no real delay between it
09:30.06P4C0maybe a small delay...
09:30.20trixternot more than 3 seconds
09:30.21trixter:P
09:30.29P4C0no more that one...
09:30.54_Lyfe_or, a song that drives everyone nuts: the "It's a small world" song by disney
09:31.10trixterheh
09:31.52P4C0hehe
09:31.59benjkCathay Pacific use to have this "Its a kind of Magic, It's a kind of Magic, It's a kind of Magic" and it went on and on and ond
09:32.00P4C0or the barney one
09:32.25benjkand there was nothing magic at all about their customer service
09:32.34P4C0hehe
09:33.01_Lyfe_the thing that always got me was repeating advertisements for services that I'm calling about not working.
09:33.43P4C0lol
09:35.04P4C0guys... here it is about 4:30 AM... and I haven't sleep, and I have to be at work around 9:00 AM... so I think that I'm going to leave... I don't want to be on job hold... :) c u later felows
09:35.13*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
09:35.24_Lyfe_ok, enjoy.
09:35.44trixterhaha barney theme song followed by the tellytubbies theme song
09:35.47trixterthat would just be mean
09:36.09_Lyfe_i think you'd have nasty letters written to you.
09:36.26oriontknand do to a firmware exploit in <SIP phone of your choice> you can never hang up!!! ever.
09:36.47trixternot even if you pull the power cord out?
09:37.05_Lyfe_err.. riiight..
09:37.40oriontknno, radio-active beta particles from an internal nuclar power convertor will continue
09:37.57oriontkndevice power if external AC is disconnected
09:37.58trixterya know I wonder how well sip devices respond to an invite they didnt initiate ...  broadvoice for example uses caller id as its authentication means once registered..  if you set the fromuser as another subscriber BV will basically talk to that persons system and if their SIP device accepts the call it will go through ...
09:38.07trixterasterisk rejects that but how many others reject it as well?
09:38.55zoahey ho!
09:39.01trixterhi zoa
09:39.10trixterwe were talking about the email I sent you a while back regardinbg the calculator :P
09:39.13trixterwell emailS
09:39.13_Lyfe_note to self: dn't use broadvoice.
09:39.15trixterhehe
09:39.52trixterlyfe: well it wont affect you if asterisk is what registers with them, you shouldnt use broadvoice for other reasons
09:40.07trixterlike the fact htey advertise they route to UK NCFA numbers but havent since global crossing dumped em
09:40.28trixterthey basically advertise falsely..  and if you cancel they will bill you overage charges..  my $10/mo account cost me $100 each for 3 accounts
09:40.32_Lyfe_they got dumped by global crossing?  now that's hard to do.
09:40.43trixtermy bank gladly refunded the money becuase I complained its merchant fraud but ...
09:40.53trixterthey got dumped cause they werent paying the full bill
09:41.05*** join/#asterisk gvag11 (n=g@ipa77.4.tellas.gr)
09:41.10_Lyfe_hah.. that makes sense.
09:41.10trixterfor a year there was a dispute on what was charged, and after 12 months GC finally dumped em and said they were sueing
09:41.22*** join/#asterisk h3x0r (n=h3xor@64.192.116.16)
09:41.24trixterfurther 'broadvoice' is a registered trademark of broadcomm *not* the clec that owns the company that owns broadvoice
09:41.26_Lyfe_that's pretty sad though, global crossing is usually has a really low price.
09:41.46trixterthe same 3 guys are directors of the clec, broadvoice, and the intermediate company..
09:42.06trixterI think the reason they didnt use their own clec for calls on broadvoice was becuase they were playing access charge/recip comp games
09:42.09_Lyfe_hmm
09:42.21ComputerWarmquestion i get -- Started music on hold, class 'default', on SIP/4034432835-c966 but i don`t hear anything. i am just trying to play a mp3 atm
09:42.30trixterits cheaper to use your own network than to buy someone elses generally speaking ...  why buy someone elses?
09:42.58_Lyfe_becuase you don't actually own your own network?
09:43.00trixterComputerWarm: I dunno ...  it works for me
09:43.10trixterlyfe: they do though, they are a clec in multiple states
09:43.18_Lyfe_weird.
09:43.37_Lyfe_i simply decided to deal with norlight communications.. they've been good to us sofar.
09:43.39trixterbut they went with a bunch of random companies to do DID support instead, my thought was that they were playing games
09:43.53trixterand if you port a number into broadvoice they own your number you cant port it out unless they let you
09:43.58trixterso if its a cool number they wont let you
09:44.22trixterthat is in their TOS
09:44.39_Lyfe_did you find that out the hard way too?
09:44.53trixterthat company has real issues, and the fact that they lied to customers, falsely advertise their services, etc, doesnt make it better..
09:45.00trixterno I never ported anything in but I did read their tos
09:45.10_Lyfe_hey BBB, go do something about this co! :P
09:45.22_Lyfe_they in the US?
09:45.46trixteryeah near boston
09:46.00trixterbut meh there are already complaints on virtually every voip board out there about them
09:46.14trixtereverything from dtmf not working to poor audio quality ot ...
09:46.15_Lyfe_got screwed by them?  contact the better business bureau.
09:46.32trixterhell 1 month after I canceled my account they had a RTP stream (2 actually) to my box
09:46.32_Lyfe_bbb.gov, i think.
09:46.39trixterI got all my money back so I dont care
09:46.46trixterinfact I profited off them but that is a different story :)
09:46.58trixterbbb is not a gov agency
09:47.19_Lyfe_really?
09:47.34trixterreally its a private group
09:47.35_Lyfe_well, damn, it's not.
09:47.37trixterand its bbb.com
09:47.38_Lyfe_how about that.
09:48.10trixterits a scam in effect, pay us money and we will list you as a good company and have money to investigate complaints about you and other companies
09:48.28trixterand any company that pays enough never gets that bad of a review, becuase everything was 'resolved satisfactorially'
09:48.36_Lyfe_heh
09:51.34_Lyfe_hey, how about that, they are listed in the BBB database already.. only 1 complaint though.
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09:53.29benjkwhat a day
09:53.31trixterthey must not have paid enough :P
09:53.50benjkyeah I was just thinking that myself
09:54.58benjkbeen doing Asterisk support on three different chats in English, in French and int halfduplex Germanglish and didn't get paid anything :-)
09:57.04trixterI m,eant broadvoice to remove all the complaints against em in  the BBB :P
09:59.03benjkwell, I didn;t think you were talking about the same thing I did, but I found it funny that you said "they didnt; pay enough" just when I was thinking that same line ;-)
10:01.31trixterheh
10:02.08benjkbut Germanglish is really tiring
10:02.21*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
10:03.26trixterich keine spreche deutch, und nicht seir gut!
10:03.48_CRC_benjk: and support to an aussie :P
10:03.51benjkyeah something like that
10:03.59_CRC_benjk: I think I have my SRV records setup properly now too
10:04.02benjkbut it was halfduplex
10:04.16benjkCRC, great stuff!
10:04.39benjkthe chap was asking in German and I was answering in ENglish
10:04.42*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
10:04.45_CRC_so, if I understand it right, if people just call crc.id.au - it should call through to the default incoming call context....
10:05.08benjkI think you need a full SIP URI though
10:05.14_CRC_ie the same if it was an enum call
10:05.16_CRC_really?
10:05.25benjkthat is you need something before the @
10:05.29_CRC_I don't have anything to test it with :|
10:05.40benjkbut then again, I never tried without
10:05.45_CRC_I thought that if nothing was there, it would go to the equiv of s@ etc
10:06.20benjkwell, from the viewpoint of Asterisk this would make sense yes
10:06.22benjkBUT
10:06.44benjkSIP URIs and their handling wasn't designed specifically for asterisk
10:07.01_CRC_true
10:07.25benjkso I would be surprised if the SIP designers had though about something like a SIP device that has anything like a default context
10:07.50benjkso my hunch is that SIP wants a full SIP URI
10:08.20_CRC_possibly
10:09.40_CRC_I don't have anything that I can test it with apart from my own asterisk setup - which may skew the results
10:10.09benjkwell, I only have asterisk too
10:10.24benjkyou can always read the RFC
10:11.03benjkI think the section on SIP URIs and the requirements how they have to be formed and all that syntax stuff is right at the beginning
10:15.54trixterthe concept of contexts is beyond the scope of sip
10:16.39trixterjust as the concept of mailing lists is beyond the scope of smtp
10:17.02trixtersip is a transport method, contexts are how the data is handled at one endpoint
10:17.45benjkyeah sure, but it is nevertheless possible that the SIP RFCs allow you to send a call just to a naked domain (without the user@ prepended)
10:18.18benjkeven though I think its unlikely
10:18.25trixterI dont think so, sip takes some of what it does from the concepts of smtp
10:18.45benjkthe To and From fields and contact etc that would all be messed up
10:18.57trixterand one of the things is requiring a user to select which destination within the sip server to route the call to
10:19.54trixtercrc is of course free to read the rfc (I have only read parts of it as needed) but I am 99% certain that you have to have a full sip uri to send to otherwise its not compliant, anyone that doesnt have all the info is nonstandard
10:20.15trixterwhich is a nice way of saying that someone may have written such an application but ...
10:20.43trixtera general rule of the internet is also to be very liberal in what you accept and very strict in what you send -- to ensure compatibility
10:20.50trixterso it would violate that doctrine to send without the user part
10:21.32benjkindeed that's what I meant
10:23.52*** join/#asterisk langals (n=icechat5@196.7.14.183)
10:24.41_CRC_which RFC is it? 3261, 3515, or 2543?
10:25.46*** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net)
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10:28.33trixteruse the highest one available, unless you have a reason to use a lower one
10:31.36trixterso tivo works with the ipod now, letting you download videos to your ipod ...  who is gonna make tivo work with a video voip call :P
10:39.29oriontknhow does tivo connect to the ipod? via itunes?
10:39.45trixterI dont know there was a news story about it the other day
10:40.01trixternot wanting to pay for overpriced electronics I didnt pay much attention other htan it could
10:46.28trixterugh back to legal stuffs, contracts are so boring
10:51.04bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
10:51.50manydejadejadejadejadejadejadejadejadejadejadejadejadejavu
10:51.54trixterthat didnt work the last 10 times I dont think its gonna work now :(  it might but I dunno
10:52.05manymh. de ja  ja de.
10:52.11manyvu.
10:52.12trixteralso you may have better luck actually stating what the price is in your automessage rather than just saying 'contact me'
10:56.28zobiahello everyone how can i get a outbound call status?
10:57.55trixterbefore, during or after the call?
10:58.56benjkmany: whats this German Japanese stuff about?
10:59.26trixterwould that be germanese or japaman?
10:59.35benjkhahaha
10:59.42benjknot its dejaese
10:59.59trixterdejaese sounds like a salad dressing
11:00.06trixteror a cheese
11:00.28benjkwell there's always jadeese
11:00.29trixter"We have a nice plate of brie and dejaese"
11:00.42benjksounds delicious
11:01.05benjkwill that be served with a nice bottle of Beaujolais?
11:01.42zobiahello. can i relate a CDR record to a call file?
11:03.13trixteryou can but they would be cousins
11:03.31trixterya know not quite the same but  they cant marry either
11:03.31lmeis anyone already tried * & digium cards on HP servers ?
11:04.13trixterso who has the asterx foam beer can holder?
11:04.19trixteror is that just me?
11:04.38benjkcan you use that with Guinness?
11:04.57trixterthose cans, at least here, are thinner so it wouldnt work well
11:05.07trixterplus they have the shaker thing so you really shouldnt drink from the can
11:05.10benjktoo bad
11:05.24trixterwell this is also a foam holder from the 80s
11:05.31trixter:)
11:05.40benjkyeah those widgets in the Guinness cans are great, arnet they
11:05.56trixterapplix had a trademark on asterx groupware stuff back then
11:06.11benjkapplix
11:06.14trixterI think they let the trademark lapse in the 90s though, but applix is still around so maybe not
11:06.19benjkblast from the past
11:06.33trixterthey were a giveaway at a NPUG meeting in the 80s
11:06.48*** join/#asterisk RoyK (n=roy@80.239.107.70)
11:07.07trixterit was either the one at disney florida or shreveport lousiana
11:07.13trixterI forget it was so long ago
11:07.23benjkwell, I think anyway, the Guinnes wouldn't go all that well with the brie chees
11:07.30trixterhaha
11:07.38trixterirish stout with french cheese, no prolly not
11:07.49trixterguiness in america tastes like water :(
11:07.54benjkdepends
11:07.54trixterwith just a hint of dog piss
11:08.07benjkif it is cheese from Brittany, it might
11:08.25trixterbrie is a soft spread that normally has a floured covering on it
11:08.31trixterthe covering always makes a mess cause the flour gets everywhere
11:08.37benjkCelto-Irish brew with Celto-French cheese
11:08.47zoathere is no floured couting
11:08.58zoaits white, thats true
11:09.03zoabut its shiny, not floured
11:09.04trixterthere is this thick layer that has a dust on it then if its not flour
11:09.13zoathere is no dust on it :p
11:09.21trixterthere is everywhere I have had it
11:09.26benjkits mold
11:09.26trixteryou peel it back to get at the cheese
11:09.33benjkmold not dust
11:09.53trixterhttp://www.the-crane.com/images/cheeze/brie-de-meaux-aoc.jpg
11:10.01benjkyou want smelly
11:10.05trixtersee on the top?  there is a fine white powder on every block I have ever had
11:10.20benjkha, wait until you've smelled a Munster cheese from Alsace
11:11.11trixterwhat I really want is this 3 day snarfing of web data to end
11:11.30benjkhttp://www.teddingtoncheese.co.uk/acatalog/de299.htm
11:11.42trixterthe dumbfucks wrote a really stupid webapp..  it takes longer the higher up in a data set you go, that tells me that htey select * from the database then filter it in their webapp going a page at a time until they finally display something
11:11.51*** join/#asterisk mutilator (n=animenod@65.111.201.79)
11:11.53trixterthe database would be fairly constant if it were doing the sort and limit
11:12.28benjkthat's a reason to have a piece of Munster with a bottle of Riesling
11:12.53lme!!!
11:12.54trixterI bet that the person that did the database integration didnt know about limit in their sql line (which works if the database orders the output) so they didnt use it..  gah I am over 20,000 pages in this one set and its taking forever
11:13.14lmejesus !!!! munster & riesling ?!
11:13.35benjklme: whats wrong about that? perfect combination
11:13.51benjkthat's how the people in Alsace have it
11:14.07lmedamn
11:14.07*** join/#asterisk roulduke (i=nnrbmyly@p508D2AF8.dip0.t-ipconnect.de)
11:14.25RoyKyummy. waffles
11:14.25benjkif it's a special day, it may be Gewurztraminer though
11:14.40benjkno, cheese
11:14.45trixtermy dad makes a riesling like wine
11:14.51benjksmelly cheese at that
11:14.57trixterhe makes a variety, that is just one of them
11:15.08lmeat least "vendange tardive" for riestling
11:15.09benjkAh, that's an honest statement "Riesling like"
11:15.12lme-t
11:15.32trixterits not from that region of germany
11:15.42trixterthe difference between champagne and sparkling wine is there for the same reason
11:15.42benjkmost vendange tardives in Alsace are Gewurztraminer
11:15.53lmeyes
11:16.00lmebut riesling is better i think.. less sugar
11:16.11benjktrix: actually that is not quite correct
11:16.31trixterwasnt reisling the wine that has to be from a specific region of germany or its not really riesling?
11:16.31benjkChampagne is sparkling winde from a region called Champagne
11:16.36trixterright
11:16.39benjker wine
11:16.46trixterif its not from that region its not champagne
11:17.00benjkNo, Riesling is the name of the grape variety
11:17.03trixterjust like with reisling if its not from a specific region in germany its not reisling, its instead reisling like :)
11:17.12benjkno
11:17.12trixterhrm which one is it then that is from the region
11:17.14trixterpeis porter?
11:17.21*** join/#asterisk squid2 (n=squid2@ppp216-186.lns1.adl2.internode.on.net)
11:17.25benjkRiesling is the name of the grape
11:17.44benjkall wines in Alsace are named after their grape varieties
11:17.56trixterahh yeah its piesporter that is from the specific region
11:18.03trixterwell fine he makes a piesporter like one too :P
11:18.12benjkRiesling, Pinot Blan, Pinot Gris, Pinot Noir, Gewurztraminer
11:18.17squid2bonjour
11:18.35benjkZeroconf!
11:18.37squid2dammit i found my way here for nothing :(
11:19.05benjkthat's cool, you saved the fare then
11:19.38trixterthe best however was when I was told at the wine shop that barbera is an italian grape only and there are no california wines that use barbera
11:19.55benjkthat's bullcrap
11:20.01trixtershows the age of the guy at the wine shop ...  cuase 20 years ago barbera was common in CA its just starting to come back though
11:20.17benjkmost grapes in Europe today are actually from California
11:20.21*** part/#asterisk squid2 (n=squid2@ppp216-186.lns1.adl2.internode.on.net)
11:20.33trixterwell 20 years ago it started to fade, it was really really common longer ago than that cause on my year of birth my parents got a bottle of barbera and on my 18th we opened it
11:20.41benjkbecause there was an epidemic in Europe that wiped out much of the vines
11:20.58benjkthey had to import new vines from California to rebuild
11:21.06trixterthe french should shower more and the vines wont die off from their stench :P
11:21.22benjkbad joke
11:21.34trixterbarbera is my moms favourite ...  afaik my dad has never made a batch of barbera
11:21.54benjkMy favourite Italian is Cannonau
11:21.56trixterbet some of my neighbors have though
11:22.17benjkits a wine from Sardegna
11:22.31trixterone of hte major businesses in the immediate area, aside from generic farming (cows, grains, etc) is grapes and wine
11:22.37benjkyou have to open it one day before serving
11:22.41trixterthere is a winery every 500 feet
11:22.54benjkfir iut to develop its full flavour
11:22.59trixterI laugh at people that open it and leave it in the bottle
11:23.02trixteryou need a proper decanter
11:23.18benjkyeah
11:23.21benjkso your'e in Oregon?
11:23.21trixternot enough oxygen gets in through the narrow neck of wine bottles
11:23.25trixtercalifornia
11:23.27benjkor Claifornia
11:23.33benjkok
11:23.49benjkthey make nice Pinot Noir in Oregon
11:23.53trixternear sacramento but far enough away that its very rural, the whole county I live in only has 40,000 people
11:24.24trixtermy dad doesnt have an ATF permit so he cant sell his wine, at most he can give small quantities away but that is it
11:24.41benjkATF?
11:24.48benjkFirearms?
11:25.00trixterits completly iunsane how does that affect interstate and foreign commerce?  he gets local grapes makes it in his house ...  where is the constitutional hook to let em license that?
11:25.02RoyKhttp://www.ronchidicialla.it/ENGLISH/main_eng.html
11:25.04trixterthe A is for alcohol
11:25.23RoyKthey produce approx 5000 bottles of that white wine per yeaar
11:25.26trixteryou gotta have a permit to sell alcohol, but as a federal agency tey dont have legal footing to collect that tax
11:25.32benjkand the F is for firearms
11:25.55trixterwe have mountain lions up here as well as other stuff so firearms arent an issue :)
11:25.55RoyKwhere are you guys from?
11:25.58benjkonly Americans can come up with such a nonsensical combination
11:26.11*** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca)
11:26.35RoyKhttp://san.siberia.net/photo/lj/234/c49ysgnw5b3aqcjg9c.jpg
11:26.47trixternext time you see an ATF agent (really a taxcollector more than anything) be sure to ask "What brandy goes with a HK mp5?"
11:26.58trixterdont be suprised if he response "That depends on what you have been smoking"
11:27.04benjkin France alcohol would never be associated with firearms
11:27.35trixterATF was department of treasury originally (now homeland security) and their goal was to collect tax on alcohol, firearms and tobacco
11:27.39trixterthat was it they were tax collectors
11:27.42lme& why not ? it's kill as much people
11:28.05trixteralcohol kills more people than firearms but then so do vehicles
11:28.11trixtervehicles kills more than alcohol
11:28.15*** join/#asterisk jgonzalez (n=jgonzale@187.Red-80-26-93.staticIP.rima-tde.net)
11:28.17jgonzalezhi
11:28.18*** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de)
11:28.21trixterbut no one is out there trying to ban vehicles saying 'think of the children!'
11:28.25benjkbut at least with alcohol you dont get shot
11:28.32jgonzalezwhat about H324M support in asterisk  ?
11:28.33trixterI get shots all the time when I drink
11:28.34benjkits a voluntary thing
11:28.39jgonzalezis supported ?
11:28.53manybenjk: it was just something from yesterday.
11:28.55shmoozlife kills people
11:28.58benjkH324M?
11:29.05jgonzalezis possible do a gateway using UMTS videocalls? (H324)
11:29.08jgonzalezyes
11:29.09manydeja deja dejavu everytime that dude spams for his phones, and everytime an additional "deja" gets added.
11:29.11benjkwhazzat?
11:29.26*** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) [NETSPLIT VICTIM]
11:29.30jgonzalezbenjk, UMTS, 3G-video-calls
11:29.32trixteryou are several off then on your deja count
11:29.40trixterbecuase its been going on for quite a while
11:29.43benjkAh, H.324
11:29.46many:-/
11:29.59jgonzalezyes, this is what i said :D
11:30.01trixterafaik asterisk doesnt have an addon for h.324
11:30.06trixterbut  then I havent really looked into it
11:30.07*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
11:30.07benjkno
11:30.15benjkyousaid H324
11:30.21manyi did sleep, i hope thats nuff as excuse :)
11:30.24jgonzalezyes
11:30.42benjkwithout the dot, its difficult to recognise amongst all that aclohgol
11:30.52jgonzalez...
11:31.22fenlanderjgonzalez: not at the moment, but it is something on my todo list...
11:31.43jgonzalez:(
11:31.47jgonzalezwhat a pity
11:31.47benjkmany: we thought you were talking about some dejaese or jadeese (Garman-Japanese gibberish)
11:32.03jgonzalezi would like to use it, asterisk works very well ..
11:32.05jgonzalez:(
11:32.30jgonzalezfenlander, and in cvs there isn't anything ?
11:33.30fenlanderjgonzalez: not that I know of - there is a project page someone created on the wiki, but I've not seen any progress
11:34.01fenlanderjgonzalez: it's something that I'd like to build, but would need funding
11:34.25RoyKzoa: ping
11:34.43benjkjgonzales, you have just been appointed as chief fund rasier
11:35.03jgonzalezxD
11:35.34fenlander:)
11:35.57zoapong,
11:36.01zoajust got a message from davy
11:36.11zoaaye aye sir
11:37.29trixterbenjk: kohii
11:37.32trixterfix that alcohol problem
11:37.51benjkallohol problem? me?
11:38.08trixteryou said earlier it was hard to read the difference between what was typed and something else becuase of it
11:38.39benjkno I said H324 was difficult to recognise without the dot
11:38.57benjkthat's a dot problem
11:39.04trixteryou said alcohol [sic] in the same sentence :P
11:39.18benjkyeah, to make it more interesting
11:39.29trixterheh
11:39.35benjkactually I didnt; say alcohol
11:39.45*** join/#asterisk RipperFox (n=ripperfo@ripperfox.beavermedia.de)
11:39.51RoyKzoa: :)
11:40.23benjkI said "aclohgol"
11:40.37benjk:-)
11:40.47trixterthus my addition of [sic] :P
11:40.51benjkdon't know what it means though
11:40.51trixterthat is why I added that
11:40.53trixter:D
11:41.26trixtergah I am only 44% done with this contract..  its mindnumbing
11:41.27shmoozhmmm walmart is selling flat tv/dvd combo for $90 on black friday :0
11:41.28benjkbut it seems to be great word
11:41.44benjkyou gotta give me credit for that
11:41.45trixterthere is a website that is supposed to have all the leaked black friday prices
11:42.03trixterI gotget what it is though..  was on cnn and many of the larger retailers like walmart are pissed
11:42.07trixterer forgot
11:42.18trixterhow did I go from forgot to gotget?
11:42.33benjkmaybe we should trademark that word
11:42.35trixteroh well anyway..  many retailers are looking at how the leaks were caused and whether or not they can sue over it
11:42.40shmoozf and r are beside g and t
11:42.54trixterI normally dotn make that mistake
11:43.06trixterI have one hand and that one is usually dead on, its the other arm that misses from time to time
11:43.22shmoozdont drink and type
11:43.37trixterso for a really sick image think about this, I have been typing with just one hand all night!
11:43.40benjkgotta find someting to use it for fisrt though
11:44.01shmooztrixter whats your right hand upto ?
11:44.02benjkcant juist trademark a word and not have a clue what it is for
11:44.22trixteroh I was in some movie theater once and during the previews one of the trailers had a character ask 'do you know what its like to type with one hand' I replied 'yes' loud enough for people around me to hear, my friends thought it was funny not sure that anyone else got it though
11:44.29trixterI dont have a right hand, never did
11:45.20benjkyeah some days I feel like having two wrong hands myself
11:45.59trixterI am however trying to build a prosthetic hand but so far nothing is going to be workable
11:46.03trixterDNI should solve that though
11:46.16shmoozso is there any user friendly web panel to add sip phones to asterisk that work out of the box ?
11:46.33trixterasterisk@home comes with amp preinstalled its a web front end to do just that
11:46.59shmoozmy friend said it still doesn't work for adding phones
11:47.00trixteryou dont have to have a@h to use amp however
11:47.10trixtermy friend added sip phones with it a few months ago
11:47.16shmoozI wrote one like more than a year ago, php/javascript works like an app
11:47.21trixterI know this becuase he had no idea about anything relating to conf files
11:47.24benjkwhy don;t you add them by hand then?
11:47.39*** join/#asterisk _are_ (n=are@62.112.159.81)
11:47.44_are_hi
11:47.50trixterhi
11:51.01*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
11:51.08Chotaireare there any problems with iaxtel lately? I get UNREACHABLE
11:51.27benjkChotaire: lately?
11:51.31_are_I seem to miss the correct word for a decent search. I look for some isdn adaptor i can plug into an asterisk server so i can attack a 'normal' ISDN phone to it. I think I remember this is not possible with most cards. Is ...
11:51.32_are_there a list of cards that support this somewhere?
11:51.39benjkhas it ever been different?
11:52.07langalsHi guys....I am using IAX2 softphones built on the IaxClient library that are dialling into Meetme conferences on Asterisk 1.0.9.....sometimes they softphones won't connect and the following error is given on the server: "Max retries exceeded to host 195.2.6.76 on IAX2/bob@195.2.6.76:28649/1 (type = 6, subclass = 11, ts=150033, seqno=43)". This happens fairly erratically. Does anyone have any idea what the problem could be?
11:52.27Chotairebenjk: what's currently the biggest iax2 provider?
11:52.39benjkGlobal Crossing
11:53.15trixterhrm does asterisk work with any linux compatible (isdn4linux) isdn card?
11:53.59*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
11:53.59trixtercause I just realized that I have a bri card from a while back with pots port ... had totally forgotten about that card
11:53.59_are_trixter: according to the asterisk page: yes. at least as 'client'
11:53.59benjktrix: only in half duplex mode though
11:54.14trixterI have no intention of using this, its an older spell caster card
11:54.24Chotairebenjk: hm ok I might have asked wrong. what's the biggest free iax2 community that I should peer with?
11:54.32trixterbut someone else might want it, maybe I will donate it to sacaug.org
11:54.43Chotairenow that iaxtel seems down for ages, I also cannot dial US and dutch tollfree numbers anymore. anyone allows this for free?
11:54.46RoyKtrixter: yes, but i' rather use an hfc-pci card with bristuff
11:54.52trixterchotaire: fwd
11:54.58Chotaireand fwd is more stable than iaxtel? ;)
11:55.03benjkChotaire, not sure, IAXtel perhaps, FWD perhaps
11:55.27benjkChotaire: Voipbusters
11:55.45trixterroyk: well I bought this in 1998 or so for personal use, at the time it was the best thing going for price/performance
11:55.53benjkfree to US, Netherlands too
11:55.57trixterI had totally forgotten about it until just now though
11:56.35trixterbenjk: arent you gonna be giving away free voip to pstn in 30 countries for like a $5/mo donation to the macsterisk project?
11:57.01trixterok I go smoke now, bbs
11:57.04RoyKtrixter: you can get an hfc-pci card for EUR 20 or so
11:57.10Chotairebenjk: sounds nice, but does it also call tollfree numbers in US/NL?
11:57.15trixterroyk: not in 1998 you couldnt
11:57.18benjkyup once we got our linkup to that telco in place
11:57.26benjk;-)
11:57.28RoyKtrixter: it isn't 1998 anymore
11:57.35Chotairelike 800/866/877/etc...
11:57.41benjkRoyK, No?
11:57.42trixterum that is why I said I would donate the card to sacaug.org instead of trying to sell it
11:57.58benjkwaht year is it then? 1999 already?
11:59.56trixterthere poof the card is donated, and I didnt even have to get out of my chair..  sacaug.org has possession already :P
11:59.59trixterthat was easy
12:00.12trixterof course sometime I am gonna have to goto my basement and actually look for and find the card but meh
12:00.42trixterits easy to donate when you organize the group that is getting the donation :D
12:00.48benjkif its that simple, I donate a can of Guinness
12:00.56benjkcheers
12:01.31benjkits still ion my fridge, but as you said that's mere detail
12:02.31benjkChotaire: US toll-free is toll-free, I tested it
12:02.55benjkDutch should also be but I don;t know because I didn't test it
12:05.53trixterheh
12:06.16trixter800-call411 will let you connect to anyone listed in the national registery free
12:06.19trixteror at least they used to
12:06.30trixterso if you have toll free access and want to call someone that is listed ...
12:06.39trixterand now there are providers that will list voip users so that is becoming easier too
12:06.40trixter:)
12:07.10Druken411 is a huge money maker for telco's
12:07.10benjkI wonder whjere this will lead to
12:07.14trixteryou just gotta sit through an advertisement to place the call, but meh its 'free'
12:07.17Drukencall it from a payphone and it's free
12:07.36shmooz911 was a huge money maker for bush and cheney
12:07.46benjkanother 10 years and we will get paid for making phone calls
12:07.52trixter411 is often outsourced so the telcos themselves dont usually make the money
12:08.01trixterits the 3rd parties that provide 411 services that make the money
12:08.10trixterbenjk: I already have done that :
12:08.12trixter:)
12:08.22trixterbroadvoice was great was making $1k/mo off each line I had
12:08.42trixterthen they got into a pissing match with global crossing and bleh
12:10.11trixterthat reminds me I need to get after someone about the wire transfer they owe me for making phone calls...  thanks
12:10.35Drukenshit, i want in that deal...
12:10.57lmesamn hp server !!!! gniiiiiiiiiii !!!!
12:11.00lmedamn
12:11.24*** join/#asterisk zotz (n=zotz@24.231.47.168)
12:11.28trixterdruken: well I do own a clec so I get special treatment when it comes to phone calls :)
12:12.28trixtersell DIDs
12:12.45trixterthe FCC in america ruled that a voip provider can get dids directly from nanpa without being a certified clec
12:13.11Drukeni'm canadian... so that has no bearing on me...
12:13.15benjkwhere do I sign?
12:13.17trixterand they are free from nanpa, although you have to be registered in BRRDS which is about $35/first year less after that if you go through a 3rd party, and odds are need SS7 becuase of number pooling
12:13.31trixteryou can still get dids from america and sell em
12:13.40trixterbenjk: ask nanpa.org
12:13.57benjkI need SS7 anyway
12:14.08trixternonot them bet its nuestar.biz cause that is who delegates all this
12:14.10trixterwho runs nanpa now
12:14.19trixterlibisup does it via pri
12:14.19benjkplus I need GSM/MAP
12:14.30trixterSS7-MAP is nice to have if you wanna do mobile stuff..
12:14.48trixterlibisup is a replacement for libpri and supports afaik all the same features
12:14.49benjkit is a requirement if you want to do ZEBRAR
12:14.55benjkZEBRA
12:15.44trixterCNAM and LIDB are nice to have if you are doing serious telecom apps
12:15.52trixterwhichi s a regular SS7 thing
12:16.18trixternormally its $0.0025/query well if you are a clec doing interconnection, accudata does LIDB at least I dunno what they charge
12:16.39trixterverisign also lets you interconnect to (according to them) all north american telcos for SS7 and they support SS7-MAP
12:16.49Drukeni'd say your talking a lil above my head... :)
12:16.49benjktrixter can you setup and run an SMSC ?
12:16.50trixterthey even have SIP7 which is a SIP IM interface to the SS7 network
12:17.29trixterbenjk: yes, but odds are for low volume its cheaper to interconnect via someone over the net
12:17.53benjkthat doesn't give you the ability to receive though
12:18.12trixterverisign lets you do that via a vpn to regular SS7 (you are on your own to have agreements to access people off hteir network but they give you the physical connection ...) there are sms specific entities that let you do everything from pay per message services get your own short code etc
12:18.43trixteryou can receive through the companies I was looking at becuase it was all set up to be a content provider bidirectional sms
12:18.53*** join/#asterisk ennuyeux72 (n=ennuyeux@host-83-146-53-34.bulldogdsl.com)
12:18.56trixterI dont know their fees though, but its all internet driven so you dont have to have gear all over hte place
12:19.24benjkwell, lets first do this service we discussed, and then we talk about some other stuff
12:20.31benjka receive SMS on Asterisk service could be big
12:20.34Drukenwhat is the best ss7 implimentation in asterisk ?
12:21.05benjkprobably SS7box
12:21.23Drukenhehe went to look at that, cannot find server
12:21.47benjk:-(
12:22.00trixterI would look at libisup
12:22.05trixterit has the most acceptance so far
12:22.08trixterespecially in europe and asia
12:22.23benjkdoes it have type approval?
12:22.36benjkETSI
12:22.38Drukendon't you need the commercial lisence for that ?
12:22.46trixterhttp://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+LIBISUP
12:22.54trixterno you need to buy it from the guy that wrote it
12:22.56RoyKbenjk: sangoma has already done their first ss7 call with ss7box and asterisk
12:23.02trixterits under the same licese as ABE but its not the same as ABE
12:23.26benjkRoyK I know
12:23.30RoyKok
12:23.40benjkI was the one who introduced Doug to the guys
12:24.02benjkhad to push him even
12:24.27benjkthey didn't really grasp at first that SS& would be an interesting spopt to be
12:24.35benjker SS7
12:26.17Drukeni'm a lil confused, what is the advantage of 227 over basic pri ?
12:26.21Drukener.. ss7
12:26.39RoyKshit. in .no you gotta pay something like EUR 30k to get registered as an ss7 zone
12:26.43benjkthat you can interface directly to the big guys
12:27.10RoyKDruken: with ss7 you route with operator prefix
12:27.14RoyKit's like BGP, really
12:27.24trixterss7 is used for many things, the most common is call setup, as a normal end user you dont need it
12:27.30trixterbut if you want to interconnect with a carrier you will
12:27.35Chotairethanks for the info, benjk.. i just found some very interesting sip provider ;)
12:27.58benjkSIP?
12:28.05benjkdont you mean IAX?
12:28.41benjkChotaire, don't forget to sign up with our service when it goes live
12:28.50benjktwice the destinations
12:29.25trixterwhen you make a normal telephone call, lets say los angeles to new york, your local provider will check via SS7 to see if you are authorized to make the call, the PIC on your line etc..  it will assign a channel for its part, hand call data to your PIC (unless you used a dial around code) and they will assign a channel and hand it to the terminating lec who checks to see all of the above (authorization to call, whether the line is busy etc).  onlyafte
12:29.25trixterr a circuit is locked down, all authorization is verified, etc then the phone rings
12:29.31trixterSMS are also sent via SS7
12:30.01trixterdrunken: in short, normal users dont need to even know what SS7 is in any way :)
12:30.15*** join/#asterisk littleall (n=littleba@cm52.epsilon173.maxonline.com.sg)
12:30.20littleallhello
12:30.41trixterafaik there are only 3 companies that make SS7 firewalls too, which is kinda sad because that network is very vulnerable to many attacks and carries a lot of critical stuff on it
12:30.54benjkdruken, its probably less interesting for people running Asterisk servers and do VoIP
12:31.35trixterSS7 was largely designed for very few but well trusted companies to connect via it, that isnt the landscape anymore, there are thousands of not so well trusted companies ...
12:31.44benjkbut if you have some specialised applications, say if you wanted to use Asterisk as the basis for a network element in a GSM network
12:31.48Drukenbenjk: i find it intresting... but at this point and time i don't need it...
12:31.49*** join/#asterisk steff (n=steff@80.125.254.220)
12:31.56steffhi all
12:32.02RoyKtrixter: ss7 has nothing to do with firewalls
12:32.02benjkthen you need to have an SS7 stack on that Asterisk box
12:32.04trixterbenjk: I am actualy working on interfacing a GSM BTS into asterisk through hte abis interface
12:32.16*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
12:32.20trixterroyk: reread what I said instead of constantly trying to correct me
12:32.23benjkand for GSM you ned to have GSM/MAP on top of the SS7 stack
12:32.30benjktrixter: COOOOOOL
12:32.45benjkwe should really talk
12:32.57*** join/#asterisk ful|work (n=fulgas@s3.http-tunnel.com)
12:32.58steffanyone solved chan_misdn.so: undefined symbol: ast_load ?
12:32.59benjktrixter I want to build an XLR
12:33.17trixterits for 2 projects, one is to make a device for some governmnet agency that came to me (becuase I was open with my explanations on the GSM network weaknesses) and the other more mainstream project is to make your handset work transparently with an IP PBX
12:33.38trixterI found a place that sells BTS units for 300 EUR which isnt bad
12:33.50trixterstandard abis interface so t1/e1
12:33.52littleallhi, for dial command : Dial(Zap/g1/${EXTEN},40,m), how to disable the 40 seconds? I don't want to specify timeout
12:34.02*** join/#asterisk coppice (n=chatzill@168.155.17.210.dyn.pacific.net.hk)
12:34.14littleallDial(Zap/g1/${EXTEN},m) doesn't work
12:34.24benjkThewre is this company in the UK they have a nanaBTS
12:34.37trixtermost of hte nano/pico BTS are more money than a rela one for some reason
12:34.41benjknanoBTS
12:34.50lmelittleall: just in case.. had you tried 0 ?
12:34.50DrukenLittle-L: uhmm.. remove the 40 ?
12:34.55trixterI looked at a 200mW picbTS that was $36,000
12:35.00trixterer pico
12:35.05trixterand it didnt really do everything I needed
12:35.06trixter:(
12:35.07lmelittleall: or ,,m
12:35.15benjkyes because they have a self tuning/self organising mesh system
12:35.16littlealllme, let me try
12:35.28trixterbenjk: this one didnt
12:35.30coppicebenjk: does 8 sound lucky in japanese, like it does in chinese?
12:35.48benjkyes
12:35.54benjkbut also 3 5 7
12:36.02benjk4 is very bad news
12:36.05trixterHmm 3 priumes and 8
12:36.08trixterinteresting
12:36.12trixterer primes
12:36.14coppice4 sounds like dead in chinese
12:36.22trixter4 has 2 ways
12:36.31trixtershi which means death as well or hon
12:36.32benjkyes same in Japanese
12:36.35trixteror was it yon?
12:36.40benjkshi = 4 = death
12:36.45steffhi, which version of asterisk work well with chan_misdn ?
12:36.46coppicelots of prices in .jp seem to be filled with 8's like they are in china
12:36.51trixteryon is my final answer
12:37.06mutilatorya
12:37.06benjkyon is Japanese Japanese, shi is Chinese Japanses
12:37.08trixterin okinawa for example they only say yon not shi for 4 for that reason
12:37.15mutilatorsame in english
12:37.16mutilatorshe = death
12:37.20zoaitch ni san chi go rok
12:37.20trixterha
12:37.24zoaitch atch ?
12:37.32trixterzoa: well close :P
12:38.00trixterin japanses ch never appears without an i..  so ichi  6 is roku
12:38.05zoayes, just like your description of cheese :p
12:38.11benjkcoppice: most Japanese words have a Japanese (kun) reading and a Chinese (ON) reading
12:38.16trixterbut sometimes you dont say certain vowels that much..  like du desu ka sounds more like du des ka
12:38.39trixtersometimes you say some longer..  tori means bird but torii is a gate thing (like the big red one at kyoto) you say the ii longer than the single i in tori
12:39.05coppicekanji sounds really tough. in chinese a few hanzi have multiple pronounciations, but it sounds like in japanese most do
12:39.36trixtersome of the base writing is the same but spoken its different..  sun, moon, base numbers etc are all the same written
12:39.55benjkyes at least one kun reading and at least one ON reading
12:39.56trixtergah you guys are distracting me from work :P
12:40.07benjk:-)
12:40.15benjksonotori desu
12:40.21zoafind work as an irc moderator!
12:40.27zoatsss
12:40.31benjkcoppice still in JP ?
12:40.39coppicenope.
12:40.52trixterwhat was funny though, just a side comment, I spoke and read japanese better than my exfiancee whose mother was a hiroshma survivor..  my ex didnt wanna learn any period..
12:40.54benjkAh ok
12:41.44coppiceI was in hiroshima prefecture this week, and I survived
12:42.01zoadid you feel any radiation ? :)
12:42.07benjkcoppice you will get cancer in 200 years
12:42.12zoayeah
12:42.14zoai also think so
12:42.15coppicethere was an earhquake
12:42.23mutilatorya know i posed this question once and no one answered me
12:42.24mutilator..
12:42.24zoaoh no, even more radiation
12:42.29benjkthat was nothing
12:42.38benjkjsut 3.9
12:42.45benjk3.9 is peanuts
12:42.46mutilatorya know how on maps large cities are usually colored yellow
12:42.53zoaits like a cell phone vibration
12:43.05coppiceeveryone seemed excited about it next morning, but I was asleep
12:43.14benjkhehe
12:43.15mutilatorwell what did maps of hiroshma look like after the bomb?
12:43.18RoyKcoppice: hi
12:43.18trixterthe lobby of the UN has a statute from either nagasaki or hiroshima back half is all melted the front is perfectly fine
12:43.37RoyKcoppice: in that document 'what's in spandsp' it looks like there's some jb stuff there. is this so?
12:44.13littleallhello, i used dial(xxx,,m) to dial out. i want to be able to hear the ring of the destination phone when the destination phone is connected (but before answer). How to do that? Currently, i can only hear the music.
12:44.13benjkthis IRC client is having a memory leak
12:44.23coppiceRoyK: kind of, but it not much use at the moment. wasn't Nix supposed to have a good one for donation?
12:44.23benjkI need to quit and come back
12:45.26*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
12:45.50puzzledmorning
12:45.50coppicesometimes I have to break off to take a leak, but my software seldom does
12:47.06*** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp)
12:47.22benjkAh that;s much better
12:47.22*** join/#asterisk DrJES (n=macleajb@TradeMart-2.EDnet.NS.CA)
12:47.28benjkwhat did I miss?
12:47.41trixterlittleall: isnt m for music on hold?  isnt R for ring?
12:47.47trixterin your dial line that is
12:48.00littleallm for music
12:48.04littlealland it works.
12:48.25trixterbut you said you dont want music you wanted a ring
12:48.30trixtermy suggestion was to change it
12:48.32trixterhttp://www.voip-info.org/wiki-Asterisk+cmd+Dial
12:48.37littleallR: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff.
12:48.44benjkSir Elton John got married
12:48.44littlealltrixter, what does this line mean?
12:48.57benjkmarried to a bloke that is
12:49.06littleallwhat is kapejod's bristuff?
12:49.11coppicethey have gay marriages in the UK now? :-\
12:49.18trixterits a patch that got added about 1.0.7
12:49.19benjkseems like it
12:49.24trixterwhat version of asterisk are you running?
12:49.44littleall1.0.9
12:49.51littlealland te411p card
12:50.09trixtergay marriages are a good thing..  think of it this way, there are about 50% men 50% women (not quite true but close)..  if a bunch of men are gay and all that is less competition for the women!!
12:50.15trixtermaking men in higher demand
12:50.18trixter:D
12:50.35Pete_Largowhat about gay female marriages?
12:50.38trixter1.0.9 should have bristuff in it, although I am not 100% certain on that, odds are though it will
12:50.45trixterPete_Largo: fun to watch?
12:51.03Pete_Largotrue, but fewer females in circulation as well...
12:51.03trixterand the ones that arent, would you really want to date them?
12:51.07coppicewhat about miserable female marriages?
12:51.08mutilatorplus if they keep up the trend they'll eventually die off again...
12:51.16benjkgay marriages have bristuff in them?
12:51.22trixterhahaha
12:51.24*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:51.25trixterthey have some stuff
12:51.27trixtersomewhere anyway
12:51.32trixterdunno if its bristuff
12:51.32Pete_Largodarwinism...
12:51.52littlealltrixter, how to show the caller ID to the destination phone? Currently, my system always show as private number
12:51.59benjkbastardo reello incredibilo
12:53.33trixterlittleall: that depends on your carrier, some dont pass caller id some you have to use their webpage to select on/off some accept it from your sip device
12:53.55*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
12:53.57trixteryou can try to do setCallerId() or something
12:54.07littlealltrixter, thanks.how can i check 1.0.9 support bristuff?
12:54.28trixterasterisk -V
12:54.31trixterit should say
12:54.35lmelittleall: which card are u using ?
12:54.41trixterdid you compile it yourself or use a package and if package from whom?
12:54.48littleallte411p
12:55.01littlealli compile asterisk myself
12:55.05trixterohhh thought you were doing voip
12:55.08benjkte411 works with BRIstuff?
12:55.15frenzycan someone help me with this...     -- Executing NoOp("SIP/XX.XX.XX.XX-08521678", "7777777") in new stack
12:55.17lmete110 does
12:55.20frenzyNov 25 07:55:27 WARNING[16765]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'callingcard'
12:55.28frenzyI've been getting this since I moved to * 1.2.0
12:55.38frenzyI have restart *
12:55.39littlealllme, how to check?
12:55.41benjkso you can use the PRI card for BRI?
12:55.41frenzyall the time
12:55.44trixterfrenzy:  exten => t,1,noop(this is a timeout)
12:55.47RoyKcoppice: dunno about nix, but zoa is working on a jb for me
12:56.17lmebenjk: i have te110P & quadbri from junghanns working together one with pri_cpe signalling other with pri_cpe_ptmp
12:56.22trixterfrenzy: you can set  the timeout if you need to alter the default
12:56.38lmelittleall: excuse me, i was not here at the beginning... How to check what ?
12:57.01trixterbristuff does more than just bri
12:57.09frenzytrixter: exten => 7777777,1,Dial(SIP/7777777)
12:57.10lmebenjk: * 1.2.0 bristuffed 0.3.0-PRE1
12:57.14frenzythats how i've set my extensiosn
12:57.20littlealllme, i mean is it possible to check the system log and find out whether bristuff is supported
12:57.26trixterok back to work
12:57.42lmelittleall: show version on * console
12:58.00littleallAsterisk 1.0.9 built by root@mobmeee.localdomain on a i686 running Linux
12:58.05littlealllme
12:58.13lmelittleall: then u'r not bristuffed
12:58.19littleallwhy?
12:58.26frenzytrixter: ?
12:58.27*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
12:58.33trixteryeah?
12:58.48littlealllme, according to what ?
12:58.51lmelittleall: 'cause you'll see somthing like : Asterisk 1.2.0-BRIstuffed-0.3.0-PRE-1
12:59.01littleallok.
12:59.08frenzyhow do I set the time out ?
12:59.29benjkso you mean I could use a t100 card for BRI if I cannot get a BRI card?
12:59.40lme!!!
12:59.47littlealllme, anyway, what is the function of bristuff?
12:59.57benjkthereby connect Asterisk to a BRI circuit using a T1 card
13:00.01lmeno :) my te110P is connected to a T2 line, my quadbri to 4 T0
13:00.17benjktoo bad
13:00.22Pete_LargoT2 line?
13:00.31lmeT1 / french version
13:00.43RoyKe1
13:00.52Pete_LargoT2 = 6M or 4xT1 ?
13:01.07RoyK?!?!?
13:01.09trixterfrenzy: http://www.voip-info.org/wiki-Asterisk+cmd+DigitTimeout
13:01.12RoyK.fr doesn't use E1?
13:02.39zoathey do use E1
13:02.39littleallIt seems that bristuff can work for 1.0.x because some one posted it in the voip-info "I have compiled asterisk/libpri/zaptel 1.03 with the patches from bristuff-0.2.0-RC3, but not the kernel driver. I used the driver provided by Vihai ( http://www.orlandi.com/zaphfc/ ) (thanks for the help!). "
13:02.53zoazaphfc sux
13:03.02zoaeuh
13:03.05zoabristuff sux
13:03.09zoanot the idea though
13:03.16lmeRoyK: E1 'xcuse !
13:03.16zoabut the way that it breaks zaptel
13:03.26frenzyI'm getting NoOp's when dialing extensions
13:03.52lmelittleall: but... why do you need bristuff if you do not use bri card ?
13:04.04macTijnhmm
13:04.09benjkbecause he can! :-)
13:04.13lmeyes
13:04.17lmeit's a fact :)
13:04.21macTijnare there sarge .deb's of asterisk-1.2.0 ?
13:04.21littlealllme, because i am using music on hold. And i want to use R option
13:04.29*** part/#asterisk DrJES (n=macleajb@TradeMart-2.EDnet.NS.CA)
13:04.32littlealllme, R option of dial command
13:04.51lmelittleall: don't work on my E1 line between pabx and *
13:05.09littleallmusic on hold?
13:05.10lmelittleall: simplyu don't care about it, i've got no sound
13:05.15lmelittleall: ring
13:05.21macTijnlittleall: that comment you just pasted was mine, and is acutally horribly outdated :)
13:05.22*** join/#asterisk fneto (n=fneto@200-232-192-168.dsl.telesp.net.br)
13:05.32littlealllme, I want to hear the ring instead of only music
13:05.42*** join/#asterisk backblue (n=moo@87-196-6-110.net.novis.pt)
13:05.52fnetoHi all, there are someone from Brasil here?
13:05.54lmelittleall: what's facing your T411 ?
13:06.38*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:06.43littlealllme, i want to hear the ring once the destination phone is connected . I use MoH because i want to hear the music before destination phone is connected.
13:06.50fnetoI'm having problems to connect to GVT and I'd like to know if somebody here had already did it!
13:07.35littlealllme, signalling=pri_cpe
13:07.54_CRC_is there any reason why ${CALLERIDNUM} would return the caller ID name, not the number.
13:08.15lmelittleall: what's at the other side ? pabx ? telco provider ?
13:08.18littleallwhat i expected is: dial and hear MoH-->connected-->hear ring-->answer
13:08.28littlealllme, telco provider
13:12.04lmeLittleall: weird that you do not hear ringbacktone
13:12.12lme... great...
13:12.20frenzyWhats the correct way to point extension 1111 to SIP 2222 ?
13:12.22_CRC_heh - he's been trying to get that going for ~3 days now...
13:12.44BrianR___grr... Aparently if a destination is busy when asterisk picks up a file in /var/spool/asterisk/outgoing, it will just leave the file in the outgoing queue and eventually place the call when the line winds up unbusy :(
13:12.44_CRC_frenzy: I'm lazy - I do a Goto :)
13:12.57BrianR___Even if MaxRetries is 0 ;(
13:13.31frenzyi'm getting a bunch of NoOp
13:13.43_CRC_I don't understand why ${CALLERIDNUM} returned the NAME, not the number :|
13:14.06backblue_CRC_: because in the number field you have puted the name?
13:14.23_CRC_I don't declare the CID at all.
13:15.02backblueit returns what? CID or NAME?
13:15.10_CRC_NAME
13:15.21_CRC_ie the username of the phone used to register
13:15.27lme_CRC_: from where call is originating ?
13:15.38_CRC_Sipura SPA-841 phone
13:15.46_CRC_going into voicemail
13:15.48lmeto another sip device ?
13:15.52frenzy<PROTECTED>
13:15.52frenzy<PROTECTED>
13:15.52frenzy<PROTECTED>
13:15.52frenzy<PROTECTED>
13:15.53frenzy<PROTECTED>
13:15.53frenzy<PROTECTED>
13:15.58lmeouragl
13:16.09backbluefrenzy: ? debug?
13:16.14frenzyyah
13:16.24frenzyis that correct
13:16.38frenzywhy the NoOp ?
13:16.53lme_CRC_: had you tried to set the callerid field in the sip peer definition ?
13:17.11_CRC_I juse use: exten => 1111,1,Goto(2222,1)
13:17.16_CRC_or something like that
13:17.41_CRC_unless you actually have to dial something else outside the same box.......
13:17.51lmeyes but... is your sip device aware of it's number ?
13:18.05_CRC_lme: sorry - I was talking to frenzy  :P
13:18.42_CRC_good question....
13:18.46_CRC_It worked in the past....
13:19.15frenzy_CRC_:  Yes the SIP endpoint is connected to the same BOX :P
13:19.28_CRC_I'm using: exten => 999,2,VoicemailMain(${CALLERIDNUM})
13:19.59lme_CRC_: in your sip.conf are u using callerid per peer definition ?
13:20.15_CRC_no.
13:20.27_CRC_wonder if I did on the old config.... *checks*
13:20.39frenzy_CRC_: exten => 1111,1,Goto(2222,1) - here is 2222 an extension or SIP ?
13:20.54lmefrenzy: extension in the same context
13:21.04_CRC_eah
13:21.06_CRC_yeah even
13:21.12_CRC_or you can jump contexts as well
13:21.24frenzyI want it to go to a SIP in the same context :P
13:21.45_CRC_using goto(context2,2222,1)
13:22.02lmefrenzy: just dial(SIP/2222)
13:22.04BrianR___Hmm... Aparently /var/spool/asterisk/outgoign serializes calls with the same destination channel - in the case of a local/ channel, the same local/nnn@context can't have more than one concurrent call :(
13:22.04frenzygoto(SIP/2222,1)
13:22.06frenzy?
13:22.08_CRC_so if 1111 was to do the same as 2222, I'd go:
13:22.23_CRC_exten => 1111,1,Goto(2222,1)
13:22.35_CRC_exten => 2222,1,Answer
13:22.40_CRC_exten => 2222,2,whatever
13:22.41_CRC_etc
13:22.47frenzyI'm alrwady uising exten => 7777777,1,Dial(SIP/7777777)
13:22.52frenzyalready using *
13:23.05_CRC_.....
13:23.14*** join/#asterisk bugant (n=bugant@80.105.82.139)
13:23.15frenzyA DID is pointing to 777777 which goes to SIP 7777777
13:23.19_CRC_we're not dialing a SIP address here
13:23.21_CRC_it's a GOTO
13:23.21frenzybut I get bunch of NoOp
13:23.26frenzyfailing to get the calls
13:23.37_CRC_it goes to another defined extension/context
13:23.52frenzyok, but WE want to dial SIP here
13:23.58lmefrenzy: no problem with the peer ?
13:24.08frenzyyap
13:24.12_CRC_but you want both extensions to dial the same thing, right?
13:24.17frenzyits actually a friend
13:24.20lmefrenzy: sip show peer & sip show registry sounds good ?
13:24.44frenzyyes. SIP UA is registered
13:25.01lme_CRC_: as i understood it, he's only want to dial an sip device with whatever extension... But it's not working
13:25.01*** join/#asterisk santoshr (i=1063@203.199.110.93)
13:25.18santoshranybody has experience with fxo box and asterisk1.2
13:25.18_CRC_lme: I thought he wanted two extensions to dial the same thing
13:25.37frenzyExactly
13:25.44frenzyI get a bunch of NoOps
13:25.51frenzyAnd then I have to restart *
13:25.54*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:25.56lmefrenzy: can you originate calls from your sip device ?
13:26.08frenzyyes
13:26.10frenzy<PROTECTED>
13:26.26lmefrenzy: who is "exactly" ? _crc_ or me ? :))
13:27.04frenzyExactly = lme
13:27.05frenzyLOL
13:27.11lmedamn lag
13:27.35lmeno nat between * and the device ?
13:27.43frenzyYes NAT
13:27.46lme....
13:28.04lmesounds like rtp does not follow
13:28.10frenzylol
13:28.11frenzy:P
13:28.16frenzyI can talk
13:28.19frenzyits not with ALL calls
13:28.22frenzyI can some calls
13:28.28frenzythen I cant get some
13:28.29_CRC_lme: callerid=bleh <101> in sip.conf worked.
13:28.38_CRC_lme: I must have forgotten to move those lines over.
13:28.45lme_CRC_: cool !
13:29.09frenzylme: has started since 1.2.0 upgrade
13:29.30lmefrenzy: hum...
13:29.34santoshrfxo box not sending dtmf to asterisk 1.20 anybody notices anything like this
13:29.49_CRC_heh - the aussie guy for voicemail sounds funny :p
13:30.34lmefrenzy: & for nat you're forwarding all ports defined in rtp.conf in the 2 ways ? incoming & outgoing
13:30.45[TK]D-Fendersantoshr :  * Topic is 'Asterisk 1.2.0 has been released! -- ftp.digium.com  (RFC2833 users see bug 5780) || http://www.asterisk.org'
13:31.04steffanyone as some experiencxe with junghanns quadBRI ?
13:31.15lmesteff: i'm using it
13:31.26*** join/#asterisk without (n=dean_dav@CPE-60-226-176-32.qld.bigpond.net.au)
13:31.42frenzyrtpstart=10000
13:31.43frenzyrtpend=20000
13:31.52stefflme: i have seen she use zaptel drivers, it's right
13:32.17lmesteff: uh.... you have to use bristuff to make it work
13:32.27RoyKfrenzy: 5000 concurrent calls?
13:32.48_CRC_RoyK: lovely defaults :)
13:32.58stefflme: ok, cause i have an avm C2 and i can't get it to work in *
13:33.01santoshr[TK]D-Fender: thanks for the update.. but again repeating my question .. have u had any experience with fxo boxes and "asterisk1.2"
13:33.06*** join/#asterisk Pj_ (n=pj@fernande.happycoders.org)
13:33.23Pj_heya party ppl
13:33.52_CRC_santoshr: I think he's talking about the RFC2833 bug....
13:33.53lmesteff: if they do not develop specifics drivers for *, no chances to get it working
13:34.10_CRC_santoshr: RFC2833 being a way to send/receive DTMF tones...
13:34.12frenzyRoyK: too low
13:34.21frenzy:P
13:34.31santoshrohh ok..
13:34.36frenzy10000 - 11000
13:34.39[TK]D-Fender;)
13:34.41frenzyrealistic ?
13:34.44santoshrsorr [TK]D-Fender:
13:34.46RoyKi'd love to see that asterisk server bridging >1000 calls
13:34.55frenzy:P
13:35.08_CRC_heh - I think my little Via C3 533Mhz would have it's ass cave in at 1000 calls :p
13:35.10stefflme: ok i go to bye a junghanns, so anyone want an AVM C2? ;-)
13:35.16lmeRoyK: i hope you've got solids firemen in your building
13:35.52_CRC_the Via C3 handles asterisk and my house security :)
13:35.54santoshr_CRC_: my iphone would send h245 to asterisk but the fxo box wont.
13:36.14_CRC_via a USB based digital I/O adapter
13:36.34_CRC_but trying to find a cheapish IP camera is a bitch :|
13:36.37trixtercrc: 1000 calls doing what specifically
13:36.52_CRC_trixter: anything :p
13:37.05RoyK_CRC_: the C3 handling 1000 concurrent calls?
13:37.05trixterwell I bet it can do 1000 if its just doing sip registration for example
13:37.06RoyKthat is BS
13:37.16trixterhe said that it would cave in
13:37.18trixternot that it was
13:37.26trixterkinda the opposite meaning
13:37.35lmefrenzy: i really think that you've got a nat issue... somemisconfiguration in sip.conf or some on your nat box
13:37.44bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
13:37.53_CRC_yeah - it only handles ~3-9 devices
13:38.16benjkis there any way we can get this bbz guy banned from the channel?
13:38.17_CRC_one call re-encoding to g729 takes ~16% CPU
13:38.25Pj_I got a quick question... I got a T2 link (haven't seen it yet) and I was wondering how I would make that work with *... Can I "split it" physically ?
13:38.26trixterser on a 400MHz ipaq can do several hundred registrations, I would like to think that your c3 is faster :)
13:38.34frenzylme:  is SER (with NATHelper + Media Proxy) seems like the solution
13:38.39frenzybefore *
13:38.41Pj_And then plug it into a quad T1 ?
13:38.51lmePj_: just buy a TE110P from digium
13:39.39trixtercrc: transcoding is quite costly, so is recording to disk (monitor, voicemail, etc) and a few other things..  that is why I wanted you to quantify your statement a little more :)
13:39.48_CRC_yah
13:40.00_CRC_but it does what I need at low power/heat
13:40.09trixterif you just acted as a registrar and tossed media streams elsewhere I bet it could do 1000 :)
13:40.18Pj_lme: TE110P support 1 T1
13:40.20_CRC_hell, the entire unit runs on less than 48W
13:40.25Pj_I don't see how I could plug a T2 in there
13:40.43Pj_(T2 = 3T1 from what I've read)
13:40.49lmePj_: so we don't agree with T2 :)
13:40.49trixter4
13:40.51Pj_(but I could have read wrong :)
13:40.53trixterand 7 t2 is a t3
13:41.11docelmoHAPPY POST TURKEY DAY!
13:41.13Pj_lme: for you it's half a T1 ?
13:41.18steffis beronet and junghanns the same hardware ?
13:41.28trixterdocelmo: ok if its post turkey day why dont you drop a turkey in the post for me :)
13:41.32_CRC_what is this T1 you yanks are going on about?
13:41.44lmePj_: e1
13:41.44docelmodunno
13:41.51trixterI think someone was pulling pjs leg and he fell for it
13:42.02Pj_:'(
13:42.09docelmoMan why does Cisco's Output Interpeter take so FRICKEN LONG!
13:42.10_CRC_how can I interface an STM160 into asterisk? :p
13:42.14trixtert2s are very rare to acutaly see, normally someone gets 4 t1s instead which is the same capacity
13:42.22Pj_trixter: exactly
13:42.33Pj_Except I just switched company and they ordered a T2
13:42.37trixtera t2 is normally run over the physical link a t3 uses
13:42.37lmei love france telecom :)
13:42.44Pj_lme: exactly :)
13:42.46trixterso the wiring cost doesnt make sense
13:42.58_CRC_lme: if you don't like it, just burn it... you'll fit in...
13:42.59_CRC_;P
13:43.11trixterwell a t2 is 4 t1s framed as t1s with extra t2 framing..
13:43.17Pj_trixter: Well considering it's zi french historical operator
13:43.22Pj_They don't need to make sense
13:43.27lmePj_: so what ft call a t2 is one E1 with 31 voice and 1 d
13:43.29trixtera t3 is 7 t2s with extra t3 framing..  so on a t3 for voice you have t3, t2 and t1 framing in there
13:43.44trixterpj: this isnt north america?
13:43.56trixtersuprised the T* is used outside of there, thought all of europe used E*
13:43.58lmePj_: no way to split it, as you'll get 2 pairs
13:44.05synthetiqthere is such thing as a t2?
13:44.05Pj_trixter: it's not (and don't ask me why they call it T2 instead of E2 :)
13:44.22Pj_lme: You have experience with ft's T2 ?
13:44.22lmetrixter: in france, what the historical operator (france telecom) call a T2 is one E1
13:44.24trixterfrance is dumb :P
13:44.25RoyKi've never heard about an E2
13:44.28Pj_Ohhh
13:44.29RoyKE3 is a 32Mbps
13:44.29_CRC_hahahahhaa
13:44.33Pj_So it's really just an E1 ?
13:44.40Pj_Damn them
13:44.40RoyKE1 is 2Mbps
13:44.45coppiceE3 is 34Mbps
13:44.45_CRC_my web site comes up 1st in a search for an australian voip provider and an SPA-2000 :P
13:44.46RoyKE3 is 34Mbps
13:44.47RoyKyes
13:44.48lmePj_: not ft... but interconnecting an alcatel pabx and * with a t2 trunk
13:44.50RoyKcoppice: sorry
13:44.57_CRC_before the actual VoIP provider in question :p
13:44.58Pj_ok
13:44.59trixterahh ok, if its just a different name for something else ..  meh
13:45.13_CRC_god bless google.
13:45.47Pj_trixter: yeah, "meh" :'(
13:45.59lmePj_: trixter, yes we're dumb.... and proud of it :)
13:46.02Pj_I'm gonna ask for a tech contact there and make him spill the truth
13:46.09Pj_before I order something
13:46.39lmePj_: good luck to speek to somebody who knows what a e1 is at ft :)
13:47.16lmePj_: but for ft, a t2 is e1. 2Mbps, 32x64kbps or 31 voice & 1 data
13:47.55lmepacked with two twisted pair
13:48.08lmeand a heavy commitment bill every month
13:49.52trixterdont you normally only get 30 voice chanels out of an E1?  the last being a dummy channel and 16 or something for signalling?
13:50.03lmetrixter : exact !
13:50.11Pj_lme: At least they'll tell me "It's 32 channels !"
13:50.15trixterahh you said 31 voice :P
13:50.19Pj_then I can pray for it to work
13:50.27lmetrixter: yes but... i'm dumb :)
13:50.34Pj_lme: Yeah I wasn't there when they choose to do that
13:50.36trixtersomeone reported problems with 17-30 not being usable in asterisk recently
13:50.46trixterlike within the last 2 or so weeks I dunno what their exact problem was though
13:50.57Pj_If I had my word I would have taken an E1 at th2 or something, mci perhaps
13:51.01Pj_definitely not FT anyway
13:51.11lmei've got this problem with my alcatel... but it's the alcatel box which do the limitation
13:51.39Pj_Well, off for coffee
13:51.43Pj_thanx guys
13:52.49lme27 WCT1/0/15 Clear (In use)
13:52.50lme<PROTECTED>
13:52.50trixterI have great empathy for google now..  tihs one site to get 50,000 pages from them is taking DAYS
13:52.51*** join/#asterisk Astinus- (n=abba@213.167.111.138)
13:53.11trixteron day 3 already..  but its almost done and soon everyone can get routes and an extensions.conf ready listing of every nationas phone routes
13:53.16zoatrixter, why do they want 30.000 phones ?
13:53.21zoaeuh
13:53.23zoa50.000 pages ?
13:53.27trixterand its not a bandwidth issue on my end its just their slow ass server
13:53.43zoawhatcha doin ?
13:53.51trixterzoa: 500,000 or so different dialplan entries, every destination that I can find
13:53.57DrukenWrktrixter: hehe spidering takes FOREVER
13:54.06trixterexpanding http://www.0xdecafbad.com/Global-Numbering-Plan.html
13:54.22trixterI have 5500 or so in that list, adding about 500k more
13:54.38trixtercomplete with whether its geographic, mobile, premium, other, short code, blah blah blah
13:54.48trixterwhich carrier if that info was avialable too
13:55.03Astinus-Hello, i was wondering, if it is possible to make a webinterface for employees where they can put away information for their numbers with asterisk as PBX?
13:55.14trixterdefine 'away numbers'
13:55.18trixterd oyou mean for a follow me service?
13:55.25zoatrixter, why not get one from that 1 site that has em all ?
13:55.29zoaor ask your carrier for it ?
13:55.35zoawhy do you need to spider it from google ?
13:55.49trixterzoa: I am snarfing from numberingplans.com but I dont wanna pay the hundreds they charge to let you download the csv so I am using their free web interface :)
13:55.58zoaaaah
13:55.59zoalol
13:56.00trixterI amnot spidering from google I just have empathy for them
13:56.06zoai get it
13:56.16[TK]D-FenderPHP 5.1.0 out.... whee
13:56.33lmeand my customers still asking 4
13:56.49_CRC_googlebot loves my site:
13:56.49_CRC_7 274 3.71% Googlebot/2.1 (+http://www.google.com/bot.html)
13:57.19trixtergot a simple perl script that will take the html and make it a csv already..  so once I get the last few pages (down to a couple hundred left) I should be able to build a big massive list for everyone to use :)
13:57.58_CRC_fark
13:58.10trixterbut its anoying because its on day 3 already..  it shouldnt take this long
13:58.16_CRC_I didn't realise that my blog gets ~800Mb of traffic each month.
13:58.16trixter:(
13:58.41*** join/#asterisk nkoza (n=nahuel@209.13.206.236)
13:58.48*** join/#asterisk santiago (n=santiago@208.195.215.160)
13:58.48_CRC_last two months were over 1Gb each.
13:59.12trixtermy stuff doesnt get anywhere near that much but then I dont publish anything highly useful
13:59.14trixtertoo lazy
13:59.14zoaasteriskguru is over 1 gb daily
13:59.16_CRC_yay
13:59.18nkozais possible to collect an entire number on a dialplan step and then setting a variable to that value? maybe requiring to finish it with by pressing #
13:59.25trixterI havea ton of stuff on paper that I really do want to publish though
13:59.35zoacrc, what is your website ?
13:59.35trixterfrom the book I was writing on the non internet based threats to data security
14:00.33trixternkoza: like with read?
14:00.35trixterhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read
14:00.38DrukenWrk_CRC_: you have a blog? i've never understood those....
14:01.37DrukenWrktrixter: if your looking for canada numbering plans, the csv is available free...
14:01.54_CRC_power just gone out :\
14:02.09_CRC_and my cordless mouse receiver wasn't on UPS power lol
14:02.11nkozatrixter: tnx! that just what I was looking for.. this is new or is present on older ast versions?
14:02.21_CRC_everything worked except my keyboard and mouse lol
14:02.43DrukenWrk_CRC_: it's not powered by the usb ?
14:02.46_CRC_www.crc.id.au
14:02.56_CRC_nah - I have external power to the hub
14:03.18_CRC_as I made a ghetto USB switch with a 4 DPDT toggle switch :p
14:03.38_CRC_I guess I should go see if it's just a breaker
14:03.51_CRC_cos I'd be pissed if my UPS ran outta j00ce and it's only a breaker :p
14:04.06_CRC_tho it went on and off about 4-5 times before dying completely
14:04.20_CRC_I should start cranking music just to piss the neighbours off :P
14:04.20trixterdrukenwrk: as is all of north america, but I wanted global and didnt feel like peicing it together anymore
14:04.26_CRC_at 1:04 an in a blackout :p
14:04.37trixternkoza: its been there for a long time
14:05.42_CRC_be even worse for the neighbours if I fire up the genny :p
14:07.10*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
14:07.14zoatrixter: i think i found the complete list for free before
14:07.39_CRC_wow
14:07.45_CRC_it looks like the entire suburb is out
14:08.03*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
14:08.26DrukenWrktrixter: feel like sharing when ya got it?
14:09.04zoatrixter: i also want it, i'll use it for asteriskguru
14:10.13trixterit will be made available on my webpage once its finished
14:10.37DrukenWrkexcelent
14:10.41trixterCSV will be first, I imagine that many people want it, half of the 5500 I currently have is from astbill the other half itu, mci, and a few other sites
14:10.42_CRC_FARK
14:10.58_CRC_power has gone to my area, and the 8-9 surrounding suburbs
14:11.09trixterI am just waiting on austria to finish, its the single largest country, bigger than the US even
14:11.11_CRC_restore time is supposed to be in ~1hr
14:11.56trixterthere is almost exactly 1000 pages left, becuase its a cpu problem on their end there isnt a lot I can do to make it go faster
14:12.19_CRC_trixter: what are you actually doing?
14:12.26zoahmm, how will you avoid bad entries ?
14:12.34zoaas those things change every month ?
14:12.47trixterI am fetching webpages from them that contain about 10 entries per page to get their complete database
14:13.03trixterthey charge you for the CSV but give it free via the web, its more intensive on their end so I must say they are stupid
14:13.33trixterzoa: largely I wont, I will just let it be the way it is..  if people submit corrections I will incorporate them, as I have done with my existing list
14:15.42trixterthe hardest part of something like this is getting a base list from which to correct
14:16.02trixterby having one sufficiently large enough its easier to attract people to help with it ...
14:16.05trixteror at least it should be
14:16.06trixter:)
14:16.35DrukenWrkSHOULD be... hehe
14:16.47DrukenWrkmake a project out of it... for the voip carriers
14:16.56zoathe carriers have that list
14:17.03trixterbut if you think about open source projects, the ones that have absolutely no code base die off pretty quick, the ones that have something somewhat usable tend to attract people to help out in terms of adding features, correcting it, etc
14:17.18DrukenWrkzoa: ok.. for the want to be voip carriers.... how about that?
14:17.25zoayeah
14:17.25trixterthat same mentality should carry to something like this
14:17.27zoawould be nice
14:17.29zoai'd use it
14:17.36DrukenWrki consider myself a carrier, even tho i only have 1 paying customer
14:17.45*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
14:17.51_CRC_lol
14:17.51zoahaha lol
14:17.52trixterI consider myself a carrier even though I am a clec in only one state :P
14:18.16trixterbut I am also scottish which means 'cheap bastid' so I dont like spending money on stuff that can be free
14:18.25DrukenWrktrixter: by being a clec you are a carrier
14:18.35zoai consider myself a carrier because i have some issues and my napolean alter ego is kind of in need of a replacement
14:18.47trixterespecially when the same company is giving it away free in one format but not in the other and its trivial to transcode it into either format
14:19.09zoatrixter: probably its not legal to put that list online
14:19.21trixterit may or may not be against hteir tos I didnt check actually
14:19.31*** join/#asterisk chapeaurouge (n=chap@85.201.80.249)
14:19.36trixterat the very least it is legal for me to snarf it
14:19.36zoai live in bulgaria, i can host anything :)
14:19.47trixterand it would be nicer for them to let me put it online than to have 23509321905 people snarf it
14:20.24_CRC_oh well, I'm gunna turn this desktop off to keep power in the UPS for servers.
14:20.25trixterI did think about what you proposed in terms of checking it..  I know the last page of every country, I can do a simple 1 page fetch and see if the country data changed at all
14:20.27_CRC_back later.
14:20.38trixterto avoid fetching stuff that didnt change
14:20.52zoatrue
14:21.03trixterthe only way that the last page wouldnt change is if exactly N got removed and N got added :/
14:21.12zoaim off now, send me a copy when its ready, you have my addy
14:21.22trixterit will be posted to http://www.0xdecafbad.com
14:21.28*** join/#asterisk demetrio (n=demetrio@62.173.180.182)
14:21.31zoacheck the timestamp
14:21.39zoathere is probably some header in the document
14:21.49zoaalthough they are probably dynamic :)
14:22.01trixterits asp so the http timestamp is not reliable, as for the other stuff afaik they dont have any means to clearly identify ...
14:22.10trixterthey are selling a subscription to their service they dont wanna make it easy :P
14:25.47DrukenWrk:)
14:26.02demetrioIf I use the M() option in Dial, the executed macro cannot hangup the channel. Is this normal?
14:28.12*** join/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net)
14:29.31demetriothe odd thing is, AbsoluteTimeout works, so I could just do AbsoluteTimout(1), but this really doesn't sound neat to me
14:29.51*** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca)
14:30.29DeeJayTwoWhat's the CVS tag for 1.2.0 + all new bug fixes?
14:30.36drumkillav1-2
14:30.41*** join/#asterisk jmacz (n=jmacz@208.195.215.48)
14:30.46kannanhelllo,
14:30.47DeeJayTwook
14:30.49DeeJayTwowhat about zaptel?
14:31.03drumkillai'm not sure if zaptel was branched
14:31.08drumkillabut if it was, it would be the same
14:31.13DeeJayTwook I'll try
14:31.14DeeJayTwothanks
14:31.17drumkillanp
14:31.42kannanneed help in error msg when compiling custom kernel for slackware, while in process of asterisk installation, how do i go about it?
14:32.26lmedamn ML110
14:32.34lmei still gets some beep while calling...
14:34.33DeeJayTwodrumkilla: I filled a bug in bugs.digium.com on November 19 and saw no update on it yet...  it's about zap interface freezing sometimes when receiving a call on a zap already in use (call waiting). I'm not sure whether it's a zaptel or an asterisk issue. Do you have any suggestion?
14:34.47*** join/#asterisk Koenvi (n=kova@labs.ascom.be)
14:35.22drumkillaDeeJayTwo: if you're using Digium hardware, you can get Digium tech support to assist you
14:35.30DeeJayTwoyes
14:35.39drumkillahave you contacted them?
14:35.48DeeJayTwonot about this specific issue.
14:36.13drumkillaok, well i would recommend having them look at the problem
14:36.51DeeJayTwoit's a quad t1 card which I think is not an unpopular piece of hardware from em.
14:37.22DeeJayTwoI'll contact them..
14:37.23DeeJayTwoThank you!
14:37.37*** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com)
14:38.03drumkillanope, quite a popular card :)
14:38.12Dr_Raynothing like a cheeseburger and fries at 6:30 in the morning
14:38.34mutilatori ate 2 breathmints for breakfast
14:38.37mutilatorO_O
14:38.56mutilatoricey cool
14:39.08demetrioyou mean, chemically cool
14:39.22DeeJayTwoLOL
14:39.55trixteronly 2?  you may want a few more :P
14:40.57mutilatoryea, tummy is growlin somethin fierce
14:41.05DeeJayTwobreathmints contains 0,001% of the iron you need in a day
14:41.10mutilatori ran outta the house too fast this mornin didn't grab any turkey day leftovers
14:41.35Dr_RayI need to start takinga  multivitamin
14:43.28trixterheh today is turkey day here
14:43.36trixteryesterday was at a friends
14:43.49DrukenWrkDr_Ray: is the 6:30 breakfast or a late supper?
14:43.59Dr_Raydinner
14:44.06Dr_RayI was up all night
14:44.11DrukenWrkthen it's ok :)
14:44.20trixterI slept all night
14:44.35DrukenWrki'd personally gag trying to pack away a burger at 6:30 in the morning if i got up at like 5 or 6
14:44.51trixterwhy?  its chemically the same morning or night
14:44.57trixterit tastes the same
14:45.33DrukenWrkjust my brain... ya gotta know.. i don't normally eat breakfast, i'm up around 6 or 7 and don't usually eat till noon or later
14:45.33trixterI have pizza, burgers, etc for breakfast all the time :D  Mmm tasty
14:45.49Dr_Rayit tastes pretty damn good after being forced to eat turkey yesterday
14:46.15trixternext year bring a burger to the dinner with you and put that on your plate :P
14:46.18trixtermake everyone jealous
14:47.09DrukenWrkDr_Ray: that board your sending me, does it use a 12V relay or a 48V relay ?
14:47.10coppicewhy do people eat turkey, when so few like it? personally I rather like it
14:47.27DrukenWrki dun mind turkey...
14:47.35Dr_RayI'm not sure..
14:48.13*** join/#asterisk liran_ (n=liran@80.178.5.17.adsl.012.net.il)
14:48.33DrukenWrkDr_Ray: okie, i'll see if it works when i get it
14:48.42DrukenWrki belive i have a 48V relay
14:48.54Dr_Raychecking payphone2000.com says 12v
14:49.18DrukenWrkhmm... we'll have to see
14:49.30DrukenWrka relay is alot cheaper than a new board.. hehe
14:49.42Dr_Raywell, if you ebay it, it's not so bad
14:49.59DrukenWrknah... i ebay'd the phone and got fucked
14:50.08DrukenWrkfrom now on i buy from payphone.com
14:50.09Koenvianyone here has experience with Intel HMP?
14:50.32Dr_RayI run the southwest alines style of payphone managemnet,.. one type of boatrd only... which is why I'm happy to toss the etx your way
14:50.51DrukenWrk:)
14:50.58Dr_RayI buy payphones on ebay all the time
14:51.01DrukenWrkthis is my one and only phone, and i've yet to have it work...
14:52.01Dr_RayI think you might missunderestimate (I love that new word) how hard it will be to get the board working.. but you are stil welcome to try
14:52.24newlaren't all pay phones ISDN these days? B)
14:52.30DrukenWrkwhy would it be hard to get the board to work?
14:52.39DrukenWrkshould be plug everything in, and presto.. i would assume
14:52.42Dr_Rayrelay, hopper
14:53.11DrukenWrkwell, the hookswitch and earpeice is 1 round connector
14:53.31DrukenWrkthe relay is 2 wires...
14:53.33Dr_Rayif you look on payphone2000.com they have a bunch of ernest parts
14:53.41Dr_Raywith pictures
14:53.56Dr_Raythey are who I buy from
14:54.04Dr_Rayparts, not phones
14:54.29DrukenWrki see protel and elcotel parts...
14:55.08*** join/#asterisk }btorch{ (n=btorch@208.63.19.179)
14:55.20*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
14:56.38*** join/#asterisk lehel (n=lehel@82.79.20.17)
14:56.44lehelhello
14:59.11*** join/#asterisk roulduke (i=3kznwkib@p508D2AF8.dip0.t-ipconnect.de)
14:59.41*** join/#asterisk Junbug (i=Junbug@69.0.31.27)
14:59.52trixterwow havent heard elcotel in a long time
14:59.58Junbugdamn it i missed walmarts laptop
15:00.18trixterback when I ran the payphone company ...  97 or 98 or so I was looking at them as a supplier, but they are really expensive
15:00.29trixterJunbug: what laptop thing are you talking about?
15:00.47DrukenWrktrixter: who's really expensive?
15:01.11Junbughp 15" 256meg 40gig/wireless/burner etc.,   $379
15:01.14trixterhttp://blackfriday.gottadeal.com/Online
15:01.23Dr_RayDruken - under our products
15:01.28trixterDruken: elcotel
15:01.31trixterat least they were back then
15:01.37Dr_Raythe laptop is a lossleader to get idiots in the store today
15:01.49trixterBest Buy50% off All Fuji DVD Blank Media
15:01.52trixterthat may not be bad
15:02.08Dr_RayI run ernest instead of elcotel because I bought an enrest phone with the software
15:02.09trixteris the laptop no longer available?
15:02.23trixterthere is a rural walmart here..  and what cpu does it have?
15:02.37Junbugsempron 3000 or 2800  i forget
15:02.38trixterby rural I mean laptops wont be in high demand
15:02.50trixterthat seems close to what you get on ebay every day :/
15:03.09Junbuglol just sold on ebay for $718
15:03.26Dr_Raymy brother was eyeballing a $150 PC from  bestbuy
15:03.50Junbughmmm $150  not bad regardless of specs
15:03.57trixtercircuit city is $7 for a 50 pack of DVDs ...  it is starting to look like its time to stock up
15:04.27trixterwow cause I have seen a lot of hose types of laptops on ebay for like half that
15:04.28Junbug$7?  perfect!!!!
15:04.33trixteralthough there is an idiot born every minute
15:04.36*** join/#asterisk Querypath (n=none@medinf-25-121.fh-friedberg.de)
15:04.50trixterJunbug:  best buy is 50% off fuji DVDs
15:04.52Querypathgood morning..uhh..evening..afternoon?..whatever :)
15:04.57trixterso either way it isnt bad
15:04.59Dr_RayI hear protel is a good payphone brand too
15:05.15trixteroh and um best buy has the ATT callvantage ATA for $30 you can reflash
15:05.41QuerypathI have a little problem for quite some time now and worked on it for 2 days and still got no improvement *sigh*
15:05.58JunbugQuerypath: just give us details
15:06.04Querypathim about to..
15:06.38Querypathtrying to get an ISDN card running (hfc-s) with 1 ISDN phone on it (nt mode)
15:06.46Querypathno other isdn card present..
15:06.46trixteruhhhh
15:07.06trixtercompusa has 802.11g pci card, pccard, AP and USB adapter for $2.99 each after rebate
15:07.19Querypathafter I wrote and rewrote that zapata.conf
15:07.31Junbugtrixter: how do u see all these deals so fast?  j
15:07.37trixterhttp://blackfriday.gottadeal.com/Online
15:07.40trixter:)
15:07.51trixterthat is the webpage I couldnt remember earlier
15:08.04Querypathi still get (with asterisk -d) ERROR[2913]: chan_zap.c:10271 setup_zap: unable to load config zapata.conf
15:08.16*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:08.16*** mode/#asterisk [+o anthm] by ChanServ
15:08.25trixterwow hawking usb bluetooth adapter $10 at compusa.. that wouldnt be bad for hooking into asterisk
15:08.33Querypathand with pretty much any other command "unable to connect to remote asterisk"
15:09.22Querypathoh..by the way...running the xorcom sollution..
15:10.18Querypathany breathtaking insights?...Im about to bite into the harddrive..
15:10.55*** join/#asterisk dasenjo (n=dasenjo@63.245.87.62)
15:12.30*** join/#asterisk digime (i=digime@219.92.170.78)
15:13.18Querypathmabye chew abit on the isdn card as well...
15:15.21trixterjunbug: basically same laptop as your walmart one http://blackfriday.gottadeal.com/Item/1324
15:16.29trixter$379 toshiba celeron M cdd-rom/cdrw 15" 40gb 245MB blah blah blah ...  so you didnt miss out totally "_
15:16.30trixter:)
15:17.03zoawow
15:17.06zoathats almost nothing
15:18.59Junbugtrixter: ahh your right
15:19.09Junbugeven though i hate celerons
15:19.38Junbugwow best buys sight is really getting bogged down
15:19.43Junbugerr site
15:21.09trixterits dead on my other computer
15:21.11trixtercant get into it
15:21.14trixter'unavailable'
15:21.20DrukenWrkanyone know if a TDM FXS port will understand rotary dial ?
15:21.22*** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com)
15:21.22Junbugheh sloooooooooooooooow here
15:21.24trixtereveryone is waking up and hammering it
15:21.49wunderkinyeah.. day after thanksgiving.. time to do christmas shopping ;P
15:22.42Junbugyep.. this is when retail chains revenue numbers really count for investors
15:23.28Junbugi think walmart is goonna make a killing because they had a national wide campaign for their sales
15:24.08trixterAcer AMD Turion 64 15.4" Laptop - $699.99  http://blackfriday.gottadeal.com/Item/1045
15:24.36Junbugbah too powerfull for my xfce/debian needs
15:24.57shido6wow
15:24.58trixternot mine but then I have pegged every system I have worked on :P
15:25.52Junbugheh
15:25.57trixterCompUSACompaq Presario AMD Sempron Laptop - $549.99 AR *
15:26.27zoathose things are soo cheap
15:26.40zoawhats best, an amd sempron or a intel celeron
15:26.56shido6turion
15:27.05*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
15:27.10shido6but I cant get it to run osx yet
15:27.43trixterofficemax has 100 pack DVD (both +/-R) for $12
15:28.27*** join/#asterisk Ariek (n=Ariek@84-245-28-221.dsl.cambrium.nl)
15:28.33steffAaarrgh! i alway have this error: chan_misdn.so: undefined symbol: ast_load  :-(
15:28.51}btorch{hey guys I just finished seting up * and connecting it to my PRI line .. everything works fine for calling out from the sip phones but when I call my DID I get a msg from the telco that I'm not accepting calls
15:28.55demetriois there any way to do something while the call is bridged?
15:29.01*** part/#asterisk Ariek (n=Ariek@84-245-28-221.dsl.cambrium.nl)
15:29.06zoayeah but for low budget
15:29.13}btorch{* gives an error that there is no rule in the zapin context in the dial plan
15:29.13zoai want to buy a laptop tonight
15:29.18Junbugme 2
15:29.23zoaand i need to know celeron or sempron
15:29.24zoa:)
15:29.54}btorch{if add exten => 1480,Anwser .. and so on then it works
15:29.58mutilatoromg omg omg Mr. Miyagi died!
15:30.05nitramzoa: get an ibm x40, with the builtin accelerometer you can play neverball ;)
15:30.21nitramscnr ;)
15:30.23Junbugmutilator: he did?
15:30.27}btorch{neithe s,1,Answer nor NNNN,1,Anwser worked ... any ideas ?
15:30.31Junbugoh shit he did
15:30.32mutilatoryea
15:33.02trixterJunbug: my walmart has the laptop you missed :P
15:33.50trixterhaha google news *today* finally got wind that 'asterisk the future of telephony' was released by oreilly..  aparently someones blog entry
15:34.21asterboylol
15:34.21trixterbut still it took long enough I read it cover to cover one night a month ago and it had alrady been out for at least a month by that time
15:35.07Junbugmutilator: bummer, so much for Karate kid 8
15:35.17mutilatori kno
15:35.19mutilatori was hoping
15:35.32mutilatorthey'll get dicaprio to play in it tho
15:37.08Junbug:)
15:37.15trixterisnt it spelled di craprio?
15:37.57*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
15:38.00Junbuglol
15:41.02mutilatorwhen doin an loa to switch a number to vongage
15:41.12mutilatorto they give you a temp number to test it out while it's processing
15:41.52_DAWany here using cmd page?
15:43.05*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
15:44.02}btorch{should exten,s,1,Answer work on a context receiving calls from a PRI ?
15:44.26[TK]D-FenderI got my Polycom IP601 & 2 attendant modules :D
15:44.44file[TK]D-Fender: yay more toys to play with
15:44.59[TK]D-Fender}btorch{ : Nope.  PRI will always call an exten matching the DID dialed.  You'd need an 'i" for a catch-all
15:45.02[TK]D-Fenderyup!
15:45.15Junbugtrixter: where do u live?
15:45.34trixtercalifornia
15:45.41}btorch{I see ... so this is nothing to do with the warning I'm getting from * about ignoring switchtype and signaling
15:46.11[TK]D-FenderNo... that just means your PRI paramaters are wrong :/
15:46.26*** join/#asterisk stuartyd (n=stuart@82.152.95.1)
15:46.40stuartydHi, can abody assist with asterisk & ISDN & Call-Forwarding & UK?
15:47.10}btorch{[TK]D-Fender will that cause any problems in the future though ?
15:47.15[TK]D-FenderDamn... Polycom doesn't offer up any sample Firmware or CFG files for my new phone and I know my IP600s under SIP 1.5 would not be compatible...
15:47.23}btorch{everything seems to work fine
15:47.59}btorch{that i didn't work iether
15:48.43}btorch{I must be missing something... digium's example for the dialplan didn't work iether
15:49.34stuartydAnybody any experience with ASterisk and ISDN?
15:49.56*** join/#asterisk virgo (n=virgo@204.60.38.102)
15:50.04*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
15:50.41MRH2hi is it correct all connections to  meetme conferences get standardised on ulaw?
15:51.53MRH2if so is it possible to change this?
15:52.06demetriohow do I terminate a call that is being bridged?
15:52.37}btorch{anyone here use bellsouth T1 with * ?
15:52.46*** join/#asterisk steff (n=steff@80.125.254.220)
15:52.57MRH2as none of the callers are using  ulaw
15:54.00stuartydAnybody any experience with ASterisk and ISDN?
15:54.08lehelMRH2? why ulaw [64kbps], and not gsm [13kbps] ?
15:54.18lehelyes stuartyd
15:55.07[TK]D-Fender}btorch{ : What example of a dailplan?  Surely not "make samples" I hope....
15:55.41*** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE)
15:55.48_DAW}btorch{ I have used bell with asterisk...
15:56.30}btorch{[TK]D-Fender not that one. The had a sample for using with PRI on their website
15:56.41[TK]D-Fender}btorch{ Lets continue in PM
15:56.48}btorch{cool
15:56.48[TK]D-FenderI'll help you get started
15:57.27easimonare zaphfc and chan_capi compatible with asterisk 1.2?
15:57.36MRH2aye most callers to the conefernces are on alaw - I believe this gets changed to ulaw on the conference, I'd just like the conference to use what the majity of callers would be using instead of lots of transcoding
15:57.40*** join/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de)
15:58.00}btorch{_DAW what switch type and signaling did you use ? I'm trying some of the ones in voip-info.org but I still get the warnnings when I do a reload
15:58.10}btorch{[TK]D-Fender thanks
15:58.11[TK]D-Fender}btorch{ : PM
15:59.20_DAW}btorch{ - I am set for national switch type b8zs,esf
16:00.34virgoHello, i have been doing a lil research and cant seem to find how to create a call forward that forwards to an external phone number
16:05.20[TK]D-Fendervirgo : What kind of equipment do you use?
16:06.32virgoright now i am just making up a mock system to see if this is something we would want to give to are clients. So im just trying to configure things. I just have a box with a nic in it and an currently only uing SIP softphones
16:06.42*** join/#asterisk _CRC_ (n=CRC@gw.crc.id.au)
16:06.48_CRC_UPS didn't quite make it :P
16:06.53*** part/#asterisk Querypath (n=none@medinf-25-121.fh-friedberg.de)
16:07.08[TK]D-FenderSo for instance, someone calls and EXT and you want it to forward to say another EXT or outside #?
16:08.35trixtervirgo: I use the dial() app to call ...  for forwrding what you need is the ability to set/change the forwarding number and to enable/disable forwarding.  If you dont have an automated means (ie bluetooth presense, IM status, etc) then the database is your friend..  basically if someone has forwarding and all you just dial their number..
16:08.49dasenjoHi, I am registering an * server in a sysmaster with a sip account. I got a BYE message aprox. 80 secs after the call is established. It does not occur when I register a hard/soft phone in the same central ¿any ideas to solve the problem?
16:08.53virgoMy boss wants me to mimik are Televantage server, in the Digital Receptionist there is an option for clients to choise that forwards them directly to his cell phone
16:09.10*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
16:09.47trixterit shouldnt be that hard to do this via an agi or dbget/dbput ...  that way people can update their forwarding settings without a computer (perhaps they are on the road)
16:10.07[TK]D-Fendervirgo : All very easy to do.  Some of these things you can do on SIP phones directly, others you can just integrate into your dialplan.  All dispicably easy...
16:10.29[TK]D-Fendertrixter : I have a pre-made config file that does just that I offer up for newbs :)
16:10.39dasenjoAnother weird thing is that the problem occur only when the call in from one of the two providers .. but only with asterisk
16:14.39}btorch{<PROTECTED>
16:15.18*** join/#asterisk fugitivo (n=ajf@209.13.243.23)
16:16.01*** join/#asterisk heison (n=heison@ns.somanetworks.com)
16:18.04*** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net)
16:18.32virgoTo make like a static forward in the digital reseptionist what would i do? have to make an extention that just picks up n forwards to the cell phone
16:18.54ceph__hello
16:19.28[TK]D-Fendervirgo : Are you referring to am AMP based solution like A@H?
16:19.57virgoyea that is what i am currently trying to set up.
16:20.07[TK]D-FenderOH.... ick... can't help you there...
16:20.30[TK]D-FenderA@H does some messed up stuff.. not sure of its capabilities.
16:20.51[TK]D-FenderAnd being a GUI based frond-end to * it limits what you can do and maintain.
16:21.10virgolol, alright, do you sugjest a different method of going about this or a better GUI
16:21.45Pj_Better gui is the one you're gonna do out of frustation
16:21.46Pj_:D
16:21.51virgoIm trying to set up an asterisk system that will be equivlant with Televantages system that we can sell to clients
16:22.21virgoI dont code so i wont be creating any GUI's
16:22.30[TK]D-FenderAs a reseller?  Oh boy...
16:22.48virgobasically
16:22.52[TK]D-FenderAMP is a GUI all by itself.  if thats part of what you're looking to sell then I can't do too much for you...
16:23.30[TK]D-Fenderstraight CLI config sure... but not generated stuff like AMP
16:24.07virgook, right now im just looking to get this to work the way our current phone system does
16:24.25bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
16:24.35file[laptop]deja deja deja deja deja dejavu
16:25.53[TK]D-FenderBetter known as SPAM <-
16:30.19*** join/#asterisk Assid (n=assid@203.115.64.62)
16:30.21Assidheya
16:30.49MRH2where ru bbz?
16:31.12MRH2as u still can;t get them in the UK
16:33.52MRH2oic soz
16:37.58*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
16:40.25*** join/#asterisk lorinc (n=ang@caracas-1780.adsl.interware.hu)
16:42.21santoshrhas anybody had any experience with a h323 fxo device connecting with asterisk 1.2
16:43.21*** join/#asterisk Winkie (i=slain@cpc1-stre1-6-0-cust10.bagu.cable.ntl.com)
16:44.11Winkieguys i'm having a very weird issue, i am connecting my local asterisk server, behind NAT but with sip and iax ports forwarded, to a remote asterisk server, when people phone me it's fine, and we can both hear each other, but when i use app_conference, for some reason it doesn't see any incoming frames
16:44.18Winkietherefore i can hear everyone, but nobody can hear me
16:44.31Winkieanyone ever come across this before?
16:48.09*** join/#asterisk viLeR (i=1000@66.128.47.232)
16:50.44trixterstaples has 50 pk DVD $3 today
16:51.19[TK]D-FenderOMG
16:54.25mutilatorso.. who wants to make me a local calling area calling plan for michigan?
16:55.35sivanaif I have a sip phone and I want to grab a call from another sip phone that's ringing, is that possible?
16:56.08Winkiesivana: possibly
16:56.14iDunnoif you're in the same call group, yes.
16:56.16JonR800mutilator: www.quantumvoice.com, ask them.. they're sip only at the moment, but supposed to have IAX soon.
16:56.17iDunno*8
16:56.31sivanawhat do you mean call group?
16:56.35iDunnoif you're using a newer version of asterisk than me, you can possibly use DPickup, too.
16:56.53iDunnosivana: they're defined in the sip.conf
16:57.18demetrioanyone here knows how prepaid applications work? how do they hang up a call when credit gets zero?
16:57.28sivanademetrio: typically
16:57.38sivanaiDunno: is this stuff on the wiki?
16:58.08demetriosivana, typically.... how?
16:58.55sivanademetrio: normally thats what they do.. they beep at 1 min left then hang up with it reaches 0
16:59.31sivanawhen it reaches 0, you hang up the channel
16:59.43demetriosure, but how is that done? I mean, I'm not able to do *anything* after a Dial command until the call is hung up by other means, how can I make the cannel beep or other stuff?
16:59.47sivanaor look at the Dial() command that does it for you
17:00.13sivanademetrio: there's parameters in the Dial() that does that
17:02.28demetriohum no, there's not
17:02.28*** part/#asterisk dasenjo (n=dasenjo@63.245.87.62)
17:02.34mutilatori got a polycom 501 here
17:02.39mutilatorand i got an ip in it
17:02.40mutilatorbut i can't ping it
17:02.42mutilatorsuggestions?
17:02.48Winkiemutilator: plug it into the network :)
17:02.49iDunnosivana: yes.
17:02.51demetrioI mean, there's an options that will give a warning when x time is left
17:02.54mutilatorit is
17:02.58mutilatorit's blinking too
17:02.59iDunnosivana: and possibly in the book
17:03.00Winkiemutilator: are the link lights on?
17:03.02mutilatorhub lihgt*
17:03.08sivanademetrio: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional.
17:03.10Winkieright, what's it's mac address/
17:03.15demetriobut how do I manage multiple call from the same account?
17:03.21sivanademetrio: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
17:03.31mutilatorno idea
17:03.40sivanaiDunno: what book?
17:03.42Winkiemutilator: windows or linux?
17:03.45*** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net)
17:03.51mutilatormy pc?
17:03.54mutilatorwindows..
17:04.06iDunnosivana: Asterisk: The Future of Telephony.
17:04.06Winkieok, run 'arp /?' at a console
17:04.07demetriosivana, that doesn't solve the problem, because I still need to actually *do* something while the dial command is executed; otherwise I won't be able to manage multiple calls
17:04.09Winkiei can't remember the exact command
17:04.14sivanaiDunno: right
17:04.27[TK]D-Fendermutilator : The blinking time is because it didn't connect to a NTP time server which it needs to stop bugging you.
17:04.35sivanademetrio: no idea what you're trying to accomplish
17:04.42sivanademetrio: I answered your original question
17:05.00mutilatorthe hub light is blinking
17:05.13[TK]D-FenderOH
17:05.14iDunnohttp://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 <-- for the book
17:05.15[TK]D-Fenderhmmm
17:05.26Winkiemutilator: if it has even basic network connectivity, by trying to ping it you should get it's MAC address
17:05.29fileyay getting more specs on a project you just finished is GREAT!
17:05.32Winkiewhich is what the arp command on windows will show
17:05.47mutilatorok
17:05.50mutilatori got it's mac
17:05.56mutilatordo they not respond to pings?
17:06.00demetriosivana, right :) I had to be more specific in the first place. the problem is that what is done by L() or even AbsoluteTimeout won't help me if I need to change the timeout value *after* the call has been connected
17:06.01Winkieperhaps not :o
17:06.11Winkieif you're getting a mac it's on the network
17:06.14[TK]D-Fendermutilator : did you confirm if it took an IP?
17:06.24mutilatorya
17:06.25mutilatorit did
17:06.32Winkiecorrect subnet too
17:06.33Winkie?
17:06.38[TK]D-FenderWinkie : No, a device can report its mac regardless of whether its connected or not...
17:06.50Winkie[TK]D-Fender: no it can't, he's testing from a windows machine
17:06.52sivanademetrio: correct, you can't change it after the call is connected, and I'm not sure how that will be possible without doing it in C
17:06.58filedemetrio: you can't, what you could theoretically is have a manage thread on the box that does it... knows all the channels, the customer, and if there's multiple channels per customer then disconnect them early...
17:07.10filebut yeah you'd have to write a module to do it
17:07.14[TK]D-FenderWinkie : you check in the phones interface for what it picked up....
17:07.28demetriohmm, this sucks
17:07.30Winkie[TK]D-Fender: I know, i'm saying if he can get the MAC from the windows machine, it's on the network.
17:07.32sivanayou'd have to up the timer in the module itself
17:07.35mutilatoryep
17:07.36mutilatorheh
17:07.48mutilatori'm just trying to walk a customer through gettin it workin over the phone
17:07.53mutilatori see the mac on his router
17:07.56mutilatorbut can't ping it
17:07.59mutilatorso it's there i guess
17:07.59fileI actually have an idea in my head on how you could do it...
17:08.03fugitivoNov 25 14:06:14 WARNING[18553]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 47 scheduled tasks all at once
17:08.06fugitivowhat is that?
17:08.16mutilatorenter = .
17:08.46Winkiemutilator: does it have a web interface?
17:08.54Winkiealso anyone wanna hazard a guess at my app_conference problem? :(
17:09.03demetriofile, so prepaid applications that claim to be able to manage multiple calls from the same calling card in fact aren't?
17:09.14filedemetrio: they can handle it, sure
17:09.27fileit's just the timeout won't incorporate any calls going on
17:09.36demetriooh well.
17:10.10mutilatorhe's got a 'phone guy' on the phone with him too
17:10.14demetrioI tried to look at the source code of one of them and couldn't get how they managed to do it, so now it turns out they don't :)
17:10.19mutilatortelling him that it should be pinging
17:10.26mutilatorso *shrug*
17:10.34Winkiecould be a dead phone
17:10.48*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
17:11.55mutilatorJonR800: i mean a big old extensions.conf with all michigan local calling area
17:11.58Winkiedemetrio: what exactly are you trying to do? impliment a global timeout for an account with multiple calls so it hard kills them when the time's up?
17:12.39alephcomUsually what they do is set aside enough credit for say a 2hour call and limit the call to 2 hours.  Then the next call does the same thing out of the credit that is left.
17:13.00demetrioWinkie, I'm trying to update a timeout for the first call if a second call from the same account (credit) comes in
17:13.09filecheating really
17:13.15Winkiedemetrio: i'm just wondering if you could do it with the management interface
17:13.24Winkiebut i doubt it
17:13.36sivanaanyone here use pickupgroup=?
17:13.48demetrioactually, my idea wasn't about timeouts, but simply update the credit and hang up if no more is left, but I can't do that either
17:14.13alephcomI'm guessing it could be done but it would require a lot of work.  You would have to used something event based etc.....
17:14.32sivanademetrio: you'd need to update the timer on the channel itself.. you'll need a module
17:14.40Winkiewell if you can check and modify credit through the management interface you could write a little perl daemon to do it :)
17:14.41demetriowhat I find very strange is that there is not a way to force a call to hang up
17:14.46filethere is.
17:14.47Winkiesoft hangup?
17:14.51fileexactly
17:15.00demetriohmm ?
17:15.21Winkie*CLI> soft hangup IAX2/buttes-1
17:15.21WinkieRequested Hangup on channel 'IAX2/buttes-1' -- Hungup 'IAX2/buttes-1'
17:15.23filewhat you could theoretically do is write your own module to do timeouts
17:15.45fileand have it keep track of channels and their timeouts and information, such as the username
17:15.54alephcomexactly what I was thinking.
17:16.00fileso when you have two channels up with the same username you can effect eachother's timeouts as your own thread will be doing the actually timeout
17:16.05fileinstead of the core
17:16.05demetriofile, if there's a way to kill a call I'm golden
17:16.10Winkiefile: would it not be possible through the management interface? i'm just wondering if that's a cheeky way to do it
17:16.10filesoft hangup
17:16.16fileyou can use it in your app
17:16.22demetriobut how? via manager API?
17:16.33fileI'd just write it as a C app myself
17:16.33mutilatori can hear this polycom dude in the background
17:16.42mutilatorand he sounds like he as no idea what he's doing
17:16.47Winkiefile: i probably would too, but i don't know C as well as perl :)
17:17.03Winkiemutilator: good tech supporters are few and far between :)
17:17.04mutilator"omg something went wrong and i don't have docs for that OH NOES!"
17:17.17demetrioand if I don't want to write a C app, how do I soft hangup a call? I can't do it via AGI or in extension.conf.
17:17.39mog_homeyes you can do it in agi
17:17.47mog_homeand in extension.conf?
17:18.13demetriono, I can't, because the script is frozen until the call hangs up on its own
17:18.19trixterhaha I heard the C5 wink in pink floyd the wall and it was so quiet I couldnt tell it was my music, and I started checking my phone and stuff to see what was going on
17:18.42filetime to write app_timeout_supreme!
17:18.50trixterbut anyway plantronics bluetooth headset $10 at radio shack today, good for chan_bluetooth :)
17:19.05alephcomCheck out http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/Manager.pm
17:19.22*** part/#asterisk santoshr (i=1063@203.199.110.93)
17:19.22Winkieoh perl how i love you 8)
17:19.26fileand yes I'm seriously writing it
17:19.41demetriofile, app_kill_call would be enough for me, thanks :D
17:19.53Winkiefile: call it app_timeout_awesome please
17:20.04filekk!
17:20.26Winkie:o
17:20.41mutilatoraparently it can't dhcp properly
17:21.08Winkielaffo
17:21.18Winkieit wasn't the subnet was it?
17:21.39fileI wish we had a callback or event system that I could tap into
17:21.56mutilatorwhen he sets it static
17:21.58Winkiethat's why i thought of the manager api but i don't know much about it
17:22.08demetriook, what's a resource?
17:22.16filea resource is something that other modules and stuff can use
17:22.28demetriono, I mean SoftHangup(resource)
17:23.30mutilatorhe just wiped/reloaded it and it wouldn't dhcp
17:23.46mutilatortrying a static right now
17:24.00filedemetrio: that made no sense, try again
17:24.08fileanyway - I need to install a PSU - brb
17:24.50mutilatorshows in the arp table again
17:24.53mutilatorbut no pings
17:24.57alephcomWell, if you write it and you want to sell it, drop me a line.  I'll see what I can come up with. :-)
17:25.16Winkie:o
17:25.20Winkiesell a module eh
17:27.52asterboyWinkie: Do you know of a good #perl channel?
17:28.12Winkieasterboy: i don't i'm afraid, are you looking for anything specific?
17:28.37asterboyWinkie: just a place to talk perl as we do here to talk asterisk.
17:29.04Winkiei know there's #perl on here and on efnet, but i don't know how good they are
17:29.26asterboyWinkie: ok, I'll check out the obvious spots
17:30.22asterboyWinkie: I need to cover all the usual bases, MySQL, PHP, Perl, Python, RegEx, Java....blah blah blah
17:31.28asterboyWith all the many facets that go with this industry, its good to be connected to appropriate channels.
17:32.50fileokay back
17:37.46fugitivoasterboy: you forgot postgresql
17:38.03mutilatorand oracle
17:38.13mutilatorand C
17:38.15*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
17:38.16[TK]D-FenderAnd Dbase!
17:38.18[TK]D-Fender;)
17:38.28fugitivoand MSSQL, the best db
17:43.42*** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net)
17:44.00kippican someone help me with asterisk dieing when it loads?
17:44.21*** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org)
17:45.38fugitivokippi: run asterisk -rvvvvvvvvvvvvvvvv and see the output
17:48.06JonR800mutilator: you still need that ext conf?
17:48.20mutilatorJonR800 yea
17:48.33*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
17:48.46JonR800mutilator: http://svn.scottstuff.net/public/asterisk-lca-map/trunk/README
17:49.00JonR800there's a download link at the bottom of the readme
17:49.15*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
17:49.38Winkiewho said oracle?
17:49.42Winkiemutilator: you should be ashamed of yourself
17:49.44Winkieoh hello lilo
17:49.49fugitivonot me, i said mssql
17:49.56mutilatoro_O
17:50.03Winkiemssql is still pretty bad :)
17:50.26znoGwhy is it that a FXO card in Asterisk uses FXS signalling?
17:50.42fugitivobecause fxo receives fxs signalling
17:50.48znoGah, its what it receives
17:50.49znoGfair enough
17:51.02znoGi figured it was that, but thought i'd better double check
17:54.23*** join/#asterisk techie (i=gus@antibala.com)
17:58.46*** join/#asterisk silasj (n=silas@201-42-13-16.dsl.telesp.net.br)
17:59.39*** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net)
18:00.57LaibschMy asterisk server was running fine until I upgraded to the latest package from Debian unstable yesterday.  Now it won't start anymore although I have RUNASTERISK=yes in /etc/default/asterisk.  Output from "asterisk -U asterisk  -n -c 2>&1" is at http://rafb.net/paste/results/UvGHq743.html
18:01.20znoGjust wondering, how does caller ID work with standard signalling provided by my telco? how does it reach my end of the line?
18:01.36Qwell[]Laibsch: Thats it?
18:01.45*** part/#asterisk silasj (n=silas@201-42-13-16.dsl.telesp.net.br)
18:01.57Laibschyes, strange isn't it?
18:02.27Qwell[]take off the stderr redirect at the end, and also take off the -U and -n
18:02.31Qwell[]See if you get any more output
18:02.48Laibschbut then I run asterisk as root?
18:02.57Qwell[]Laibsch: yes, try it
18:03.10Qwell[]then gradually add the other options
18:03.17LaibschI remember that previously created problems with some files that could later not be removed.
18:03.29Laibschbecause they were owned by root.
18:03.35LaibschLet me check my mail.
18:08.05LaibschIt is http://bugs.debian.org/333351
18:08.14*** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net)
18:08.53NewSoleQuestion... I have someone complaining there system is not getting Disconnect tones from pri.... I looked but fond nothing on it
18:10.09crash3mNewSole: just duct-tape their mouth shut, problem solved
18:10.58NewSolewish I could they are biggest customer
18:11.05alephcomI like your troubleshooting skills. :-)
18:11.18IronHelixznog- its a FSK audio burst that comes after the first ring
18:11.26IronHelixif you hear it it sounds like a half second of modem data
18:11.34znoGinteresting
18:11.38NewSolewould hanguponpolarityswitch=yes be solution
18:11.42LaibschQwell[]: I did as you suggested.  Not more but less output.  Exactly nothing. -vvv also did not reveal much (at least to me).
18:13.21Qwell[]Laibsch: Are you trying to run it from an init script or something silly?
18:14.06LaibschI usually run it from /etc/init.d/asterisk but this time I did nothing but "#asterisk -vvv"
18:14.13znoGmy problem seems to be that asterisk doesn't wait long enough to detect the ring.. is there any way to adjust this?
18:14.50Qwell[]Laibsch: #asterisk?
18:15.06Qwell[]and, no, it won't do anything without the -c
18:15.10bancusthe # is a root prompt
18:15.16Qwell[](which I never said to take off)
18:15.20bancus(or so it seems to me)
18:15.42Laibschyes, I was trying to signal that I ran the command as root
18:15.54Qwell[]Laibsch: asterisk -c
18:15.56Qwell[]period
18:16.01bancusit looked like a question to me, for clarification, the only thing I could see needing clarification was the #
18:16.09LaibschGives the same output.
18:16.21Qwell[]Then it just sits there, or what?
18:16.27trixterha I got the radio shack to put the plantronics $10 bluetooth headset behind the counter for me..  at that price and limit 10 per customer I wouldnt be suprised if they dont pop up on ebay really quickly for $40 (they are a $60 MSRP headset)
18:16.44LaibschIt returns to the command prompt
18:16.49Qwell[]Laibsch: With -c?
18:16.54LaibschNo asterisk process is running
18:16.57Laibschwith c
18:17.01Qwell[]-c or c?
18:17.04Laibschwith -c
18:17.17Laibschasterisk -c
18:17.23Qwell[]This is why we don't use distro packags. :)
18:17.48Qwell[]the next step now, is for you to uninstall the debian packages completely, and install stable from source
18:18.13LaibschHm, I'd rather just downgrade.
18:18.39LaibschSeems to be the time-saving thing to do for me.  If that does not work I can always still run from source.
18:20.04Qwell[]What version is the old version?
18:21.20*** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net)
18:21.45docelmoWHADUP!?
18:21.45znoGso do you guys know how I can make asterisk wait a couple of seconds to determine the type of ring? (for distinctive ring detection)
18:22.00docelmowait(#)
18:22.16docelmo# == Seconds
18:22.23znoGbut then it would have already gone into a context
18:22.44docelmouhh well..   It should know right off the bat..   Are you using ZAP?
18:22.57znoGi have distinctive ring configured so the default pattern is mainnumber, and a dring pattern for a diff context
18:23.04znoGyep, using zap
18:23.36docelmoshould be in zaptel.conf I believe.  If you cant find it here they guys @ digium will give you support for your card for free
18:23.51znoGzaptel.conf or zapata.conf_
18:23.52znoG?
18:24.12*** join/#asterisk hadi57 (i=al_moghr@212.11.189.141)
18:24.17docelmoIf you have everything set and active then I cant help you.  I just know the theory behind setting it up
18:24.20docelmoZaptel.com
18:24.23docelmoconf
18:25.22Flautounder mandriva 2006, zaptel drivers are at a very strange location. the location is /lib/modules/2.6.12-12mdkcustom/misc
18:25.28*** part/#asterisk hadi57 (i=al_moghr@212.11.189.141)
18:25.32alephcomWhen using "T" in the dial command.  How do I specify where the transfered call goes.  I want it to hang up....
18:26.04bancuswhy not just use the hangup command?
18:26.18docelmoNot possible..  Unless you send it to  a catch all context that matches the number then Hangup is your option
18:26.40IronHelixalso znog
18:26.44znoGdocelmo: not much in zaptel.conf in regards to how many rings to wait to answer or stuff like that, just the zone and signalling
18:26.45IronHelixif you're still interested in caller id
18:26.47IronHelixtry this http://www.jungroup.com/vonage/ringing2.wav
18:27.05IronHelixits a recording of a ringing line from a very noisy ATA (vonage of course)
18:27.21znoGIronHelix: heh ok :) i was just curious as to how the number got sent on an analog line
18:27.35alephcomIt's for a calling card agi script I'm working on.  I want the user to be able to disconnect the other end and go on to place another call.
18:27.54alephcomOk, I'll see what I can figure out.
18:27.56docelmoI would say speak with Digium or consult the wiki
18:28.08Qwell[]alephcom: Perhaps you want the h or H option?
18:28.10LaibschQwell[]: The current version is 1:1.2.0.dfsg-3, the old version I am not 100% sure but i think it was 1.0.9.dfsg.1-3.4
18:28.13Qwell[]I think thats it anyhow
18:28.32*** join/#asterisk cypromis (n=michael@asterisk.pl)
18:28.45alephcomI was using it but that disconnects my end of the call.
18:29.01docelmoya.. alephcom you could use the h option to send them back to make another call when they hang up or do a goto() statement and send them back
18:29.19Flautowhen i use modprobe zaptel it tells me module zaptel not found
18:29.32Qwell[]Flauto: is zaptel installed?
18:29.39Flautoyes
18:29.53Qwell[]How did you install it?
18:30.01Flautomake clean
18:30.10Flautomake linux
18:30.13Flauto26
18:30.15Flautomake install
18:30.17alephcomok, I'll mess with it some more. tks
18:31.26Flautothe locations is /lib/modules/2.6.12-12mdkcustom/misc
18:31.54LaibschQwell[]: Did that information help at all?
18:32.04Qwell[]Laibsch: no, not really
18:32.13Laibsch:-(
18:32.18Flautoit not at /lib/modules/2.6.12-12mdk
18:32.24Qwell[]the "dfsg.1-3.4" simply means "we butchered the shit out of these packages."
18:32.24LaibschI will write a bug report and downgrade.
18:32.30docelmoYou may need to type modprobe modname to activate it.
18:32.30LaibschLet't hope it helps.
18:33.53Flautodocelmo, i was doing modprobe zaptel but it tells me zaptel not found
18:34.07Flautoeven when i did modprobe in the misc folder
18:34.16Flautoi also did depmod
18:34.18docelmoya..  Cause Zaptel is a pseudo name..
18:34.25docelmocat /etc/modprobe.conf
18:34.31docelmoyou will see the valid names there
18:35.53Flauto/sbin/modprobe
18:36.03Flautoi go this
18:36.09bancusLaibsch: actually, dfsg means that the original source had something non-free in it, and it had to be removed to be included in debian
18:36.19docelmodude..  just type cat /etc/modprobe.conf
18:36.25docelmoand look in there..  What card do you have?
18:36.41Flautoi have x100p
18:36.46Laibschbancus: Thank you for the information.  I highly value the Debian Free Software Guideline.
18:36.48docelmoHOLD....
18:37.20Flautoi see wcfxo there
18:37.28bancusLaibsch: cool, thought I'd just throw that out there, given the prior biased definition
18:37.49docelmotype modprobe wcfxo that should start it.
18:38.11docelmoYou should however consult the wiki to find the correct driver for your card.  I use the TDM410P
18:38.53Flautobut it does not
18:39.00*** join/#asterisk vlrk (n=vlrk@59.93.71.116)
18:39.01*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
18:39.33docelmodoes not what?
18:40.12Flautoi understand that i need to modprobe zaptel  and modprobe wcfxo
18:40.33Flautobut it says module zapte not found
18:40.40Flautozaptel
18:40.42Flautosorry
18:40.52Flautoi used depmod -a
18:41.05NewSolecan some one help me.....
18:41.27docelmoNewSole, Dunno?
18:41.37docelmoYou dont activate Zaptel for the mod
18:42.04docelmoyou activate your actual card's drivers..  Hold..
18:42.23Laibschbancus: yes, I was very tempted to correct it.  but what good would that do?  Qwell has also been very helpful for which I am grateful.  He is entitled to his opinion about Debian and you and I are entitled to ours.  There are times to be a Debian missionary and there are times when better to shut up.  I do not know much about
18:42.39Laibsch* so I better shut up.  And I'll do that right now ;-)
18:43.00NewSolewe have a phone switch connected to TE410 Card.... and the phone switch is not getting hangup/disconnect signal from asterisk box...
18:43.30docelmoHow is it connected?   What signaling are you using?
18:43.47NewSoleNI2....
18:43.52Flautoit is a pci card, i installed it already
18:44.09Flautoand let me check the signaling
18:44.11*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
18:44.12docelmoNot switch type..   I mean um,
18:44.15bancusLaibsch: heh
18:44.17docelmocrap
18:44.42NewSolepri_net
18:45.08docelmoya that shit..  um, whats your phone switch setup as?
18:45.27docelmoWho is master and slave?   Who is CPE and Office?
18:45.29NewSoleNI2 with D Channel
18:45.40bancusLaibsch: in the next couple months I plan on deploying a Debian-based system as a PBX, and I'll probably package my own asterisk debs if you're interested
18:45.41docelmoSo your Setup for NI2 PRI
18:46.05Flautofxs_ks
18:46.08bancusI can see why one might want to compile from source, but that's no reason to ditch dpkg
18:46.09NewSolecan we PVT... make eas to read
18:46.19LaibschWhat do you want to change?  Why not try to get the code into debian itself?  why the fragmentation?
18:46.24Laibschbancus: What do you want to change?  Why not try to get the code into debian itself?  why the fragmentation?
18:46.54bancusdebian is apparently doing the fragmenting, ATM I'm curious what's non-free
18:47.04Qwell[]bancus: the hold music
18:47.13bancusaha
18:47.18Qwell[]and, I said that, more specifically about the "1.3-4"
18:47.20Qwell[]or whatever it is
18:47.22docelmoFlauto, your driver is WCFXO
18:47.32Flautoyes, sir, i know
18:47.34docelmoNewSole, yes.. IM me..
18:47.50Flautocan i pm you? docelmo
18:48.01bancusof course, yeah, if I add things, I'd send the patches to the appropriate places
18:48.11docelmoNo..  1 at a time..  I am doing 5 things right now
18:48.13bancusbut I like having features
18:50.12bancusin particular the features that are removed to deal with patent issues (I assume that's why the hold music is taken out)
18:50.26Qwell[]bancus: no, just the music itself
18:50.32bancusoh, well, hell
18:50.39Qwell[]the fpm mp3's
18:50.40bancusdidn't realize * came with music
18:50.46Qwell[]three mp3s, yeah
18:50.54bancusI guess the ubuntu packages leave it out too.
18:51.05fugitivomeetme works like shit in 1.2 for me
18:51.34Qwell[]it's only non-free because they can only be used in asterisk.  The license for the music explicitly states that
18:51.36bancusI read some article somewhere that went through this whole thing replacing the mp3 player for MoH because it said mpg123 couldn't be used in a commercial environment
18:52.02bancuswhich is BS, because mpg123 is GPL last I looked
18:52.28Qwell[]fugitivo: got a timer?
18:52.51sylei prefer madplay myself
18:52.55sylecan control the volume etc
18:53.21fugitivoQwell[]: i'm using a x100p
18:53.30Qwell[]kram: y0
18:53.34kramsup
18:54.00Qwell[]fugitivo: Have you looked at bug 5827?
18:54.00NewSolehey kram>
18:54.09kramhi newsole
18:54.16fugitivosyle: true, i'm using native moh, and had to amplify the mp3 volume :)
18:54.20Qwell[]and/or 5697
18:54.23fugitivoQwell[]: no, what bug?
18:54.28NewSolegot a tough question for u
18:54.29Qwell[]fugitivo: look at both of those
18:54.33fugitivook
18:54.53NewSolekram> got a tough question for u
18:55.18*** part/#asterisk vlrk (n=vlrk@59.93.71.116)
18:55.52NewSolehow can I send a Disconnect Signal to Zap channel when call ended
18:55.56fugitivoQwell[]: well, good to know it's a known bug
18:56.07Qwell[]fugitivo: Is that the same thing you're experiencing?
18:56.09fugitivobut i'm not using ztdummy
18:56.22fugitivoyes, same problem
18:57.22Qwell[]fugitivo: I believe there are patches on that bug.  You should try them out, and see if they help
18:59.43fugitivoi have another problem
18:59.48fugitivomaybe related to the timer or zaptel
19:00.19fugitivoa remote user, using iax phone, calling using the x100p, his voice is really robotic and choppy
19:00.26fugitivoif he calls me to my sip phone, the voice is clear
19:01.40Qwell[]I don't think 5697 is related to just meetme
19:01.49Qwell[]it could be negotiating a different codec or something
19:02.07*** join/#asterisk hans (n=fugalh@falcon.fugal.net)
19:02.42*** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com)
19:02.56hansunder what circumstances does asterisk fork about 20 children? (out of curiousity)
19:03.18Qwell[]hans: I find it does that on my RH9 box when I use the safe_asterisk init script
19:03.42hansnot safe_asterisk here. it is 1.0.7, btw (debian sarge)
19:03.59*** join/#asterisk darkskiez (n=darkskie@host86-134-0-160.range86-134.btcentralplus.com)
19:04.17Qwell[]hans: no, I mean, the init script calls safe_asterisk
19:04.29hansI know, I checked that it's not calling it
19:04.34zoaKRAAAAM
19:04.42hansI've had issues with safe_asterisk so I double-checked that.
19:04.49Qwell[]zoa: Too late!
19:04.52Laibschlab
19:08.07znoGis there any way to show debug/verbose output in asterisk or do i have to edit asterisk.c and set option_verbose=1 ?
19:08.38iDunnoasterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv -g -c -r
19:08.49iDunno(or silimar)
19:09.57|dennis|question: I just setup an TDM04B(with 4 fxo ports) i am testing right now..i called my self adn set it up to echo anyone who calls in..the problem is when the person calling hangs up..asterisk is not detectin gthe hangup...how do i make it detect the hangup?? please help...
19:10.15wasim~docs
19:10.17jbotrumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
19:10.37wasim|dennis|: read up on echo cancellation and hangup detect
19:10.44|dennis|ok thanks..
19:10.56*** join/#asterisk bluedemon (n=shannonm@merlintechs.kvinet.com)
19:11.00wasim|dennis|: especially on the wiki
19:11.05bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
19:11.19|dennis|ok...thanks :) sorry for being silly and not doing that first...
19:12.44Qwell[]shh, he's back
19:13.24bluedemonHas anyone here had any problems with a delayed sidetone when using the Monitor app?
19:13.58kramdennis: e-mail support@digium.com
19:14.06znoGhi kram
19:14.41docelmoKRAM THE MAN OF DA HOUR!
19:15.03znoGkram: sorry to bother, quick question if you have the time: in the asterisk source, can i increase the time it takes for asterisk to determine the distinctive ring pattern?
19:15.37docelmoHay Digium Guys..   Anyone know when the dCAP information and certifications will be released?
19:17.09zoaznoG, there is
19:17.20zoathere is a config option for it
19:17.30zoabut it can cause problems
19:18.08znoGzoa: whats the conf option?
19:18.14sylethink your on your own if you case about certifications hehe
19:18.18sylecare
19:18.34zoaznoG: hmm
19:18.36zoalet me check
19:19.05docelmomeaning?
19:19.24docelmoI paid for it I would like to have it.
19:19.32docelmoMay my clients feel all warm and fuzzy inside
19:19.52syleto each his own :)
19:21.23docelmoTrue..
19:22.04zoahmm i think im wrong znoG
19:23.17zoaWhaaaaaa
19:23.26zoa67 voip providers using iax2 known to me now
19:23.29zoaits amazing
19:23.42docelmohehe..  Am I on there?
19:23.45zoa67 out of 811, thats really good
19:23.47znoGzoa:                                 res = ast_waitfor(chan, 1000);
19:23.51zoawhats your provider name ?
19:23.51znoGis that it?
19:24.06zoadunno, i was thinking about something else i think now that i read on it
19:24.09docelmoZoa whats the site for the list?
19:24.16zoawww.voipcharges.com
19:25.02zoaare you on it ?
19:25.17docelmoya
19:25.22docelmoWell Im looking
19:25.26docelmoif not I will tell ya
19:25.50Qwell[]zoa: At Astricon, at dinner at that Persian place...I didn't get a chance to introduce myself properly.  I was the guy sitting right next to you near the end of the table there.
19:26.27Qwell[]had to leave early, otherwise I would have
19:26.31zoayou are the bastard who left me alone to sit next to captain crunch
19:26.34zoawell thank you :p
19:26.36Qwell[]haha
19:26.43zoathat was not very friendly
19:26.44Qwell[]sorry :p
19:27.00zoabut i can imagine why you left :p
19:27.33sivanazoa: is that your site?
19:27.38zoahehe
19:27.40sivanavoipcharges
19:27.41hypa7iayou didn't have to sit next to him in a cramped taxi!
19:27.43hypa7ia:p
19:27.46zoahaha
19:27.58fugitivowhat's wrong with that guy?
19:28.00hypa7iathat was scary :/
19:28.00zoahe didnt shut up for 1 second
19:28.01fugitivohe smells bad?
19:28.01Qwell[]hypa7ia: what was up with you staying, anyhow?
19:28.18hypa7iaQwell[]: i had to re-arrange my flight, but i did :)
19:28.25hypa7ia<3 southwest
19:28.29Qwell[]last I heard, you were on the way to the airport, heh
19:30.08Qwell[]I don't know who it was, but we drove to dinner with some guy...who doesn't own a car
19:30.12Qwell[]THAT was scary
19:30.17Qwell[]no offense to said guy, if he's in here. :p
19:30.38Qwell[]When I said it was legal to make a right turn on a red...I meant if you look both ways first! :P
19:31.31myke420247captain crunch was at astricon?
19:31.32myke420247i'm sorry
19:31.38myke420247makes me a little less sad i didn't go
19:31.57Qwell[]he's cool...for the first few minutes
19:32.03myke420247yeah
19:32.14myke420247did he ask any of you to go back to his room for yoga exercises?
19:32.28Qwell[]umm
19:32.44hypa7iamyke420247: you were at HOPE weren't you
19:32.48Qwell[]zoa: ?
19:32.56myke420247yes
19:33.37hypa7ianono it was krisk he was picking onthis time
19:33.46zoamyke420247: damn
19:33.50Qwell[]myke420247: So, I guess that's my cue to not take him up on his offer of going to a LUG meeting with him?
19:33.52hypa7iamyke420247: so you heard about the druid / hhh shenanigans
19:33.58zoait was krisk and one of the digium people
19:34.05hypa7iaoh yeah
19:34.08hypa7iacreslin, no?
19:34.11myke420247dunno about that in particular, but i've seen enough firsthand
19:34.14zoano not creslin
19:34.23zoahe was under age, can't tell you his name :p
19:34.32myke420247i've made a point of completely avoiding him the last 2 HOPEs
19:34.33hypa7ialol
19:34.49hypa7iahe seemed less crazy at H2K4 than at astricon
19:35.03Qwell[]What did I miss exactly?
19:35.03hypa7iabut then i didn't really talk to him at H2K4
19:35.09myke420247i met him at my first hacker con in 1992 and he's been creepy the whole time
19:35.21docelmoWho?
19:35.24myke420247did you see the cool public terminal cluster at h2k4?
19:35.37docelmoMust be talking about Crunchy
19:35.45hypa7ia||<-- deep end                cap't crunch-->||
19:35.57hypa7iamyke420247: i did, and used it
19:36.03myke420247cool, that was my project
19:36.09hypa7iahehe FreeBSD95
19:36.31hypa7ianice
19:36.35hypa7iait was welldone
19:36.40myke420247i've been doing those at every HOPE, but probably not doing 2006
19:36.46myke420247cuz i'm still paying for 2000 on credit cards
19:36.49myke420247thanks
19:36.53hypa7iaouch :/
19:36.54myke420247it was probably the best one
19:38.04*** join/#asterisk jahani2 (n=k@adsl-186-44-192-81.adsl.iam.net.ma)
19:38.28jahani2what diference beteween FXO and FXS ?
19:38.36Qwell[]~fxofxs
19:38.38jbotsomebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
19:38.49docelmoFXO == From CO   and FXS == from Phone
19:39.07*** join/#asterisk diverge (n=user@cpe-65-25-16-186.neo.res.rr.com)
19:39.39jahani2thank you
19:40.07*** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net)
19:40.43zoaand now lets say that fxo = fxssignalling and fxs=fxosignalling and make sure he completely freaks out :)
19:41.17*** part/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de)
19:41.28jahani2ok
19:41.51IronHelixbut in case hes not confused yet, remember to mention that each one can also use loopstart, groundstart, and kewlstart, and that he needs to figure out the right one!
19:41.52IronHelix:)
19:42.19IronHelix(almost all phones use or will work with kewlstart, so dont worry)
19:42.43Qwell[]and if they don't, you'll only see very subtle errors :p
19:42.47Qwell[]if at all
19:42.57Qwell[]oh, AND
19:42.58IronHelixso it will just 'act wierd' and you wont know why
19:43.02Qwell[]fxos use wcfxs
19:43.15Qwell[]okay, I made that one up
19:43.33IronHelixbut if you are confused, worry not, because our support fees are very reasonable!
19:43.38hypa7iameme, a power outage toasted my dad's norstar... trying to sell him on an asterisk replacement :)
19:43.43hypa7iaerr hehe
19:44.06Qwell[]It's sad when you have to sell people on a better/freeer solution
19:44.10IronHelixyeah
19:44.12IronHelixit is
19:44.19hypa7iawell, he'd need hardware
19:44.32hypa7iaand the norstar still works, they just need to reprogram it all :p
19:44.37Qwell[]oh
19:44.41Qwell[]still :p
19:44.41IronHelixalso- why is it that nobody bothers to plug their $20,000 PBX in through a $30 surge suppressor?
19:44.43hypa7iaso, i'll prolly just put in a UPS
19:44.57hypa7iait's a $1,500 pbx, and it's my dad :p
19:45.05IronHelixi mean in general
19:45.09hypa7iai know
19:45.14hypa7iapeople == stupid
19:45.17Qwell[]hypa7ia: Let me guess...
19:45.20IronHelixits like telecom guys have never heard of surge bars
19:45.35Qwell[]hypa7ia: your dad has been in telephony for years?
19:45.50Qwell[]and you weren't really that interested in telephony when you were younger.  but then all of a sudden, you started to?
19:45.52IronHelixhyp- you might be able to program it yourself, often that type of thing is possible but difficult to program using a phone or a pc with software
19:46.00Qwell[]to be*
19:46.10IronHelixalso- make sure the batteries in the pbx are fresh, so it wont lose its memory next time
19:46.56IronHelixa client of mine (before they hired me) had a similar situation, dead backup batteries.  They were out for more than a day waiting for lucent or whatever to come out and reprogram their thing, ended up costing $1500 just to have it programmed
19:47.25Qwell[]seems like they got off relatively cheap
19:47.44IronHelixyeah they were lucky, but it wasnt anything high end
19:47.55IronHelixAvaya PARTNER with about 10 extensions and a few lines
19:48.05hypa7iaQwell[]: my dad's a lawyer :)
19:48.11Qwell[]oh, heh
19:48.15Qwell[]I was way off then
19:48.23hypa7iayup
19:48.29hypa7iaparents are super non-techy
19:48.29IronHelixended up getting htem a backup PC card so now if it dies again the programming is stored somewhere
19:48.47hypa7iayeah, my dad's assistant luckily knows how to program norstars
19:48.56IronHelixthats good
19:49.08Qwell[]hardcore
19:49.10hypa7iaand i do too if she runs into trouble :)
19:49.13IronHelixi read the programming manual for the PARTNER, looked painful
19:49.23hypa7iafeature ** config baby
19:49.28IronHelixlike you gotta take a desk set with display plugged into extension 10 port
19:49.33IronHelixpunch a bunch of wierd buttons
19:49.36hypa7ianorstar is ridiculously easy
19:49.38IronHelixthen print another overlay for the sfotkeys
19:49.48IronHelixthen like you dial #xx to access each feature
19:50.04IronHelixits decently flexible, except that you cant renumber extensions
19:50.36IronHelixbut for the low low price of only $385 they will sell you an addin card that lets you dial the pbx with a modem and upload a config file
19:50.40Qwell[]I loath the Nortel we have at work.
19:50.45Qwell[]We don't even get freaking cid
19:50.50hypa7iahaha, ouch
19:50.52hypa7iawhat it it?
19:50.58IronHelixugh
19:51.01hypa7iai think we have an option 21 in here
19:51.02Qwell[]dunno, some nortel
19:51.17Qwell[]I'm not in that dept, so I try to ignore it.
19:51.25hypa7iagood plan :)
19:51.32Qwell[]I say "just give me an analog line, so I can use my 'fax machine'"
19:51.35IronHelixpartner is wierd like that too, because they have an ivr through the voicemail module, half the time when it rings it says the call is coming from PARTERMAIL VMS
19:51.40Qwell[]by fax machine, I of course, mean asterisk box :p
19:51.45hypa7ialol
19:52.05Qwell[]I actually liked the avaya system we had at my old location
19:52.12Qwell[]rather...I liked the phones.  They weren't complete shit
19:52.17Qwell[]on our nortel...okay...fucking...
19:52.21Qwell[]we don't have a god damned mute button
19:52.27Qwell[]UNLESS we're on speaker!
19:53.05Qwell[]So, if I want to listen on a private conference call, I have to cover the mic whenever I want to talk to somebody else, or everybody in my office can hear the call
19:53.28hypa7iaoh boy, i screwed up with that once :s
19:53.39IronHelixwtf
19:53.44IronHelixno mute button
19:53.54Qwell[]IronHelix: oh, no, there IS a mute button technically
19:54.01IronHelixtechnically?
19:54.08Qwell[]but when you press it...the handset is disabled, and it switches to speaker
19:54.15Qwell[]but, it does mute that mic
19:54.18IronHelix....
19:54.22Qwell[]exactly
19:54.33IronHelixwhere the hell does the logic for that come from
19:54.46Qwell[]it's in the documentation for the phone
19:54.51Qwell[]it isn't something they can just change
19:55.02IronHelixon every phone i have ever used, there are two buttons, MUTE and SPKR.  mute mutes the calls, SPKR turns on the speakerphone
19:55.11Qwell[]there is a speaker button too
19:55.32IronHelixthats lame
19:55.37IronHelixwhoever designed that is a moron
19:55.39Qwell[]it's absolute crap
19:55.42Qwell[]Nortel did :p
19:55.49IronHelixwhat i dont understand
19:55.57IronHelixis what was the engineer who came up with that THINKING
19:56.02Qwell[]Nortel
19:56.11Qwell[]oh, what was he...right
19:56.14IronHelix"lets see, maybe mute is too simple, so lets make mute also turn on the speakerphone!"
19:56.26Qwell[]nah, more like
19:56.43Qwell[]"We can't control the handset.  That would be too difficult.  Let's just mute the speakerphone mic instead!"
19:57.02Qwell[]"but, in order to do that...let's go ahead and switch the call to speakerphone also
19:57.21hypa7iai just wish i could get a cheap handset that worked with the handset button on my ciscos
19:57.30hypa7iaerr headset
19:57.57*** join/#asterisk [bsd] (n=bsd@203.134.195.5)
19:57.57IronHelixlol
19:57.59Qwell[]hypa7ia: plantronics makes some adapters
19:58.15Qwell[]all the plantronics sets work with any type of phone, as long as you get the adapter
19:59.07hypa7iahmm, cool
19:59.07Qwell[]actually, the headset button on my 7960 is weird...I don't know if it's just that one, or if it's sccp or what
19:59.10*** part/#asterisk [bsd] (n=bsd@203.134.195.5)
19:59.22znoGim pretty sure my problem is related to the fact that asterisk used to take a while from the time it says "Starting simple switch on Zap/1-1" until it actually answers the call, thus allowing the ring style to be properly detected
19:59.26Qwell[]but if I'm holding the handset, and there is audio coming out...it's slightly quiet (because I turned down the volume), and it I press the headset button, not only can I still hear it in the handset, it gets like 3x louder
20:00.03hypa7iaweird :s
20:00.14Qwell[]znoG: That happens if you have no callerid on the line, but you have callerid=yes
20:00.30Qwell[]asterisk will see it, but it won't hit any contexts until it timesout
20:00.58Qwell[]I saw it usually take about 5 seconds for that to happen.  When I turned off callerid, it would hit right away
20:01.54znoGusecallerid=yes
20:01.56znoGthat ?
20:02.19Qwell[]yeah.  If you don't have callerid on the line, that'll make it wait until it gets it (which means it'll wait until it times out)
20:02.26Qwell[]which, in your case, MAY be a good thing?  heh
20:02.46znoGheh, well i do have caller ID, that's a problem there
20:02.55Qwell[]there should be a timeout for the dring stuff though
20:03.08znoGyeah there should be, can't seem to find it in the code
20:03.13znoGlooking at channels/chan_zap.c
20:03.48*** join/#asterisk krischna (n=chatzill@adsl-216-103-90-139.dsl.snfc21.pacbell.net)
20:04.10IronHelixznog you were trying to use distinctive ring yes?
20:04.13znoGyep
20:04.33IronHelixdo you have usedistinctiveringdetection=yes
20:04.39znoGyes :)
20:04.53znoGi'd be a real idiot not to have it set
20:04.58*** join/#asterisk jahani (n=k@adsl196-33-29-206-196.adsl196-1.iam.net.ma)
20:05.14*** join/#asterisk Seba_soy (n=s@64.76.126.29)
20:05.23IronHelixdo you also have the dring cadances defined?
20:05.23Seba_soy<PROTECTED>
20:05.50znoGIronHelix: no, just the defaults
20:06.00znoGIronHelix: oh you mean the patterns?
20:06.12IronHelixhttp://www.voip-info.org/wiki-Asterisk+ZAP+channels  scroll down to Detecting Distinctive Ring on Incoming Calls
20:06.16IronHelixdid you do all that stuff?
20:08.05znoGyes, all that
20:08.13IronHelixhmmm
20:08.14*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
20:08.15IronHelixthat is odd then
20:08.29IronHelixif you enable dring detection, it should wait 2-3 rings before answering
20:08.34Qwell[]Isn't there something that can watch the line, and it'll tell you what the dring used was?
20:08.42Dr-Linuxanybody can hell me with IVR stuff ?
20:08.43IronHelixyeah, asterisk with debug on
20:08.44znoGIronHelix: should, yep
20:08.58znoGQwell[]: yep it always detects 0,0,0 as the pattern
20:09.04IronHelixdr-linux- yes we can hell you with many things.  What would you like to be condemned with first?
20:09.05Qwell[]neat
20:09.17znoGringing either number reports 0,0,0 which is wrong, of course
20:09.27znoGthe second number should be a different pattern
20:10.10IronHelixyeah, something is wrong there
20:10.20IronHelixim thinking maybe its not getting that dring detection should be on
20:10.23IronHelixthus is answering quickly
20:10.30IronHelixdo you have the latest zaptel?
20:10.44Qwell[]"usedistinctiveringdetection" is spelled properly?
20:10.53Qwell[]quite a long option name...easy to misspell
20:11.01IronHelixhehe
20:11.14shido6constantinopenoloppe
20:11.30Dr-LinuxIronHelix: hell ? :S
20:11.34IronHelixi once tried to troubleshoot callerid for an hour, wasnt going through because i put callerid=asreceived
20:11.44shido6dudecahedR4n0rZ
20:11.46IronHelix<Dr-Linux> anybody can hell me with IVR stuff ?
20:11.49znoGQwell[]: its definately ON as it shows the distinctive ring pattern detection stuff in the log, the problem is that it reports 0,0,0 for any calls to either of the numbers
20:11.55Qwell[]oh
20:16.36*** join/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de)
20:19.54LaibschI have downgraded my asterisk package.  But still "asterisk -vvvvv -c" will not start.  The console output is at http://pastebin.ca/31227
20:20.02LaibschI did not spot anything odd.
20:20.21Qwell[]My advice from earlier still stands
20:21.02LaibschThis clearly is config issue.  The older packages did work.
20:21.18Qwell[]You just said downgrading didn't help
20:21.40LaibschI also said that it used to work, didn't I?
20:21.56LaibschWhy are you trying to take the complicated route?
20:22.26Qwell[]If you don't want to take my advice, then by all means, don't
20:22.38Qwell[]however, I'm certain that you'll get the same advice from other people as well
20:22.43*** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net)
20:22.50*** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no)
20:22.55Rawplayerwhich softphone can i use under linux?
20:23.02Qwell[]Rawplayer: sip or iax2?
20:23.08znoGx-lite, iaxComm
20:23.08Rawplayeriax2
20:23.11Rawplayerk
20:23.15Qwell[]yeah, iaxcomm is very nice
20:23.22Qwell[]nice and simple
20:23.56hansthe UI has some issues though
20:24.02Qwell[]hans: oh?
20:24.12hansthe accounts dialog, for one
20:24.15Qwell[]yeah
20:24.21Qwell[]I'll give you that one, heh
20:24.43Qwell[]it kinda bugs me that when you click dial, it takes the focus from the input box there.
20:24.45hansI can't seem to get the osx version to work for me, either.
20:25.02hansyeah, lots of little things
20:25.08LaibschHow do I compile asterisk without it completely fucking up my system?  Can I force it into a temporary directory?  Does it honour --prefix?
20:25.19Rawplayerread the makefile?
20:25.35Qwell[]Laibsch: It's best to remove the debian package entirely, and start from scratch.
20:25.40Qwell[]put your configs away somewhere, just in case
20:25.52LaibschI am just testing
20:25.56Laibschno valuable configs
20:25.57hansLaibsch: you have to hack the makefile. you can put it under an absolute directory, but that doesn't work like --prefix
20:25.59syleLaibsch: did you make sure you did a rm -f /usr/lib/asterisk/modules/* before downgrading and installing?
20:26.14Qwell[]syle: he used a package.  Debian should be smart enough to do so
20:26.33Laibschsyle: I only used aptitude
20:26.37Qwell[]if not, then...my advice still stands
20:26.56Qwell[]along with my slander towards packages
20:27.01tainted_what port does fastagi sit on?
20:27.12tainted_by default
20:27.12Qwell[]port?
20:27.27hansLaibsch: if you want to stick with the packages, then purge it (_ in aptitude) and then reinstall. save your configs first
20:27.39hansI can't remember if purging will nuke everything in /etc/asterisk or not
20:27.40tainted_is it 80?
20:27.57Laibschhans: I did a purge (p in aptitude).
20:28.00tainted_n/m 4573
20:28.20hanshm, I thought it was _. oh well
20:28.25Laibschhans: I actually manually checked that /etc/asterisk and a couple of other dirs were gone.
20:28.28Qwell[]tainted_: What does fastagi listen for?
20:28.36hansif you purged and reinstalled, then modules shouldn't be the problem.
20:28.37sylewell i did some reading on creating rpm packages yesterday and if its source it would only do what the Makefile told it to, so its possible those modules would never get removed if it did a simple make install, but if you removed the other package first it should delete them
20:28.56tainted_Qwell AGI events
20:29.03Qwell[]syle: most package creators are smart enough to not use make install
20:29.13Qwell[]or, rather, not to use cp in the make install
20:29.31Qwell[]tainted_: oh, guess I just don't know enough about agi...
20:29.34hansor at least verify that make install does what they need (and asterisk's would fail that test miserably)
20:29.35LaibschI am not smart enough.  What can i use instead of make install?
20:29.39Qwell[]I figured it would just be through pipes
20:29.42Qwell[]Laibsch: make install
20:29.49Qwell[]Laibsch: we're talking about the people who make the packages
20:29.57sylehehe i don;t trust package creators, who knows how much weed they smoked that day
20:30.18LaibschI know.  But how do I avoid make install?
20:30.21Qwell[]packages for most things are fine, but I've always found that packages of asterisk are just...bad
20:30.23Qwell[]Laibsch: You don't
20:30.24sylejust to a create a package there is a damn book on it hehe
20:30.25hanssyle: heh, most package maintainers would say the same thing about users. :-)
20:30.45hansQwell[]: why do you think that is?
20:30.49hansout of curiousity
20:30.51Qwell[]Laibsch: uninstall the debian package, get the asterisk source, make, make install
20:31.00hans(I have my own theories)
20:31.01Qwell[]hans: I've just never seen any good * packages.  I don't know/care why.
20:31.08Qwell[]I just know they never work. :p
20:31.22Qwell[]even gentoo packages, which you compile yourself, aren't good
20:31.25LaibschYou conversation did not make me trust that asterisk build system is up to speed to not ruin my entire system.
20:31.25mog_homewe move to fast...
20:31.34sylei have to agree with qwell, asterisk changes so much it would be quite a chore for package maintainers to get everything right
20:31.40hansI'll tell you why, as a debian package maintainer. Asterisk's makefiles are a big mess. it doesn't preclude good packages, it just encourages bad ones.
20:31.43Qwell[]Laibsch: we're talking about people who make packages for asterisk.  Those people are assclowns. :)
20:31.49hanssyle: that too
20:32.19LaibschI don't even know what assclowns are.
20:32.26Qwell[]Laibsch: thats for the better then
20:32.34fileyays my app is almost done!
20:32.36hansLaibsch: it won't ruin your entire system. It'll splatter things all over, but you can figure out where they were splattered and remove them.
20:32.43sylepastebin is down so i can;t even see your output hehe
20:32.52hansit's all in the makefile
20:32.55Qwell[]mog_home: Somebody calls Digium, says their debian installed asterisk doesn't work...whats the first thing you tell them? :)
20:33.13fugitivoinstall from source
20:33.20fugitivocan i work with digium now?
20:33.28syleaww now i see it
20:33.30Qwell[]fugitivo: with?  sure!
20:33.36syledid it crash right after chan_zap.so?
20:33.47Qwell[]syle: according to him, it isn't crashing.
20:33.49fugitivohehe
20:33.55Qwell[]If he's giving misinformation, then...
20:33.56Laibschsyle: I returned to the command prompt.
20:34.07Laibschsyle: It returned to the command prompt.
20:34.21fugitivoit happened to me
20:34.24sylewhats the problem hehe
20:34.31LaibschQwell: I never specifically said it was or it was not crashing.
20:34.50Qwell[]well, whatever, I stopped caring
20:34.59Qwell[]When you install asterisk from source, perhaps I'll start caring again
20:35.08Seba_soyIS MARKSTER HERE?
20:35.16Qwell[]Seba_soy: NO
20:35.35Seba_soyhe joins this channel frequently?
20:35.38Qwell[]sometimes
20:35.42Qwell[]snocetti?
20:35.44Seba_soyyes
20:35.46Seba_soysnocetti
20:35.52syleso i;m almost complete the billing module
20:36.04Seba_soyi am snocetti
20:36.06sylestarted a feature freeze
20:36.12Qwell[]Seba_soy: /msg him, he may be around, but hiding
20:36.21Qwell[]Don't just say hi though :p
20:36.32Seba_soy:)
20:36.34Qwell[]tell him why...bug number...etc
20:36.43Seba_soylet me see
20:36.54Qwell[]oh, duh...helps if you know who :p
20:36.55Seba_soy0005839  [Applications] app_dial  menor  siempre  11-23-05 14:33  11-25-05 14:33
20:38.45sylei do have somewhat of an advanced question if anyone is able to help: The dial L flag forces your asterisk server to maintain the RTP stream, is there another way to do pre-paid billing cutting off call after so many seconds without maintaining the RTP stream? even someway with the Dial L flag?
20:39.32filesyle: it'll work on SIP... as the signalling still goes through Asterisk
20:39.33Seba_soysyle, I am doing my own prepaid application
20:39.42filefor cutting the call off
20:39.46*** join/#asterisk Mjolinor (n=Mjolinor@cpc1-burn3-3-1-cust226.manc.cable.ntl.com)
20:39.47filebut for an audio warning, nada
20:40.31Seba_soydo you speak spanish file?
20:40.37fileno
20:40.41syleso whats the proper way to handle that , by just not setting the LIMIT variables for the dial L flag?
20:40.43Qwell[]file: Why not!?
20:41.07sylei heard code was broken in latest releases so that is good to hear
20:41.46sylemaybe they meant by iax not sure
20:42.00znoGthis is so strange
20:42.02znoG<PROTECTED>
20:42.02znoG<PROTECTED>
20:42.02znoG<PROTECTED>
20:42.02znoG<PROTECTED>
20:42.04znoG<PROTECTED>
20:42.07znoGwhy oh wy
20:42.10znoGwhy
20:42.33Seba_soysyle, use S option from dial app instead L option
20:42.55fugitivoznoG: do you have s,1 in your default context? :)
20:43.32sylehmm good idea
20:43.39Seba_soysyle, do not answer the call before launch Dial APP
20:43.54znoGfugitivo: that's not the prob. The problem is that as soon as the call enters, it is quickly answered and therefore the ring pattern is 0,0,0
20:44.21fugitivoznoG: oh :)
20:44.26syledoes it forward iax to? or is RTP stream still there for S as well?
20:45.02fugitivoznoG: wait command?
20:45.06Seba_soyin SIP just put CANREINVITe=yes
20:45.15Seba_soyboth sides
20:45.52sylei see, and iax?
20:46.08Seba_soyI dont know iax, I only use SIP
20:46.20syleok thx for the input
20:46.37Qwell[]iax doesn't have rtp, so...
20:46.42znoGfugitivo: wait would be executed once it has already determined which context to go to
20:46.51Seba_soy???. iax does not have RTP?
20:47.02Qwell[]Seba_soy: it's all in the same stream
20:47.10*** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin)
20:47.15PakiPenguinhello everyone
20:47.28fugitivoznoG: but it can be executed before answer
20:47.30Seba_soyso it cant be compressed?
20:47.55Qwell[]Seba_soy: compressed?
20:48.00znoGfugitivo: yes, but i need the wait to execute before the distinctive ring detection goes on
20:48.56Seba_soyi mean, iax is like ulaw, and uses a lot of bandwidth
20:48.56znoGthere's no doubt the problem is asterisk not waiting 2 seconds after starting switch. Why is the question.
20:49.12fugitivoSeba_soy: iax is a protocol, not a codec
20:49.48Seba_soybut you say that voice is inside protocol
20:49.48fugitivoSeba_soy: you can use the same codecs with iax and sip
20:50.36Seba_soyi have a question
20:51.24moralewhat would someone recommend, the linksys sipura 841 or grandstream gxp-2000?
20:51.33Qwell[]morale: looked at the spa-941?
20:51.37Mjolinorcan someone tell me how to make asterisk look at the line before it's picked up to make sure no one is on the phone from the landline in parallel to the asterisk server?
20:51.38Qwell[]linksys
20:51.47Qwell[]Mjolinor: chanisavail
20:51.53Mjolinorcheers
20:51.55Seba_soyI am making my prepaid app in C, so I answer the call, ask for a pin, then ask for a destination. There is a manner that when I launch Dial from inside My C program RTP pass directly from my CISCO GW to ENDPOINT withoyt asterisk in the middle?
20:52.01Qwell[]Mjolinor: though, it may not work, because asterisk expects to own every line it touches
20:52.14Mjolinorok, Ill google it, thanks
20:53.40Qwell[]shido6: ?
20:53.42shido6if you hand off the call how do you keep track of it?
20:53.48shido6Seba_soy
20:54.03Qwell[]shido6: asterisk could still own the signalling, heh
20:54.10Seba_soyyes
20:54.13Seba_soyexactly
20:54.35Seba_soysome like if SIP tells cisco that media have to go to another equipment
20:55.27myke420247hey has anyone had problems dialing 800/866 with voipjet or nufone?
20:56.05shido6got a valid cid?
20:56.32*** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m)
20:56.32trixterhrm aside from the insane lag that shouldnt be there, this headset seems to work ok ...  although I cant tell if the noise that I was hearing was packet loss, interference with  the wifi, 2.4GHz cordless that I used to answer the call or something else
20:56.54sivanazoa: thanks for updating my listing
20:57.00myke420247yeah, and toll calls are fine
20:57.03trixterbluetooth -> ipaq -> (wifi) asterisk -> lagged voip provider -> my cordless..  not a really good test environment
20:57.09myke420247but toll free to most numbers gets fast busy
20:57.22Seba_soyquestion, how many calls can i catch simultaneously using AGI?... (for prepaid)...
20:58.10trixterdepends on your system
20:58.18syleSeba_soy
20:58.44syleone last question for you, is it possible not handling the RTP stream could screw up your cdr records, i;ve heard reports on this
20:59.00Seba_soyp4 2ghz, memory 512, scsi disk, 3com ethernet 10/100.
20:59.46Seba_soysorry, I did not understand the question, do it more prehistorik style, I speak spanish :)
21:00.45Seba_soytrixter, I am experimenting some choopy sound on a Athlon 3000, 512 Ram, ATa disk, when it reach 25-30 simultaneous calls.
21:00.49syleby not handling the RTP stream, possible to screw up call detail records
21:01.31Seba_soysyle, all you have to got is, START and END of the call, it is signalling I think... not rtp
21:02.40Seba_soyit means, MEDIA is going from endpoint to endpoint, but signalling still going to asterisk.
21:03.26Seba_soytrixter, I want a system to handle 120 simultaneous calls for a PREPAID CALLINGCARD. what can I buy?
21:03.45shido6anyone need a pair of 7960 phones?
21:03.48trixterare you sure that its not your network?
21:03.49syleoww i heard the problem is they wont match the upstream provider at time
21:03.50Qwell[]myke420247: Are you by chance settings your cidnum to an 8xx DID?
21:04.01Qwell[]shido6: Free?  Sure! :)
21:04.04fugitivoshido6: how much?
21:04.10shido6$200 each
21:04.14Seba_soymmm CISCO is in same network than ASTERISK and are connected to a 3Com Switch
21:04.16Qwell[]not bad
21:04.23shido6"lightly" used
21:05.38trixterSeba_soy: how are the calls?  voip both in and out, pri in and out, one of each?
21:05.48Seba_soyVOIP both
21:05.53trixterwhat codec?
21:06.12Seba_soyCISCO receives call in a E1, send it to ASTERISK and it send destination call to other's ciscop GW
21:06.13myke420247ok
21:06.15Seba_soyg729
21:06.22myke420247qwell, yes
21:06.31myke420247just figured out i can't use a toll free number
21:06.36myke420247is that actually documented anywhere
21:06.36myke420247?
21:06.42trixterso you have an E1 into asterisk?  and that transcodes to g.729?
21:07.04sylehe said cisco
21:07.07Qwell[]myke420247: no, it isn't a problem with asterisk or nufone.
21:07.08Seba_soyno, E1 is connected to CISCO, cisco send incoming calls to asterisk in g729,
21:07.22Seba_soyasterisk launchs AGI asking for PIN and DESTINATION
21:07.22Qwell[]myke420247: My work does the same thing...I had to hack up my dialplan with some gotoif's based on the number I'm calling
21:07.32trixterahh ok, so its only voip on the local network?
21:07.37myke420247so you can't have a tollfree CID when making a tollfee call, or something?
21:07.37Seba_soyasterisk send call to other gw's, it depend of destination and LCR
21:07.39Qwell[]if I'm calling to 8xx, I just set my cidnum to something else
21:07.42trixteror is it voip both?
21:07.47Qwell[]myke420247: it works with some numbers...doesn't with others
21:07.57Qwell[]myke420247: It's a "problem" with either their provider, or their pbx
21:08.05Seba_soyonly VOIP, i have not other kind of equipment
21:08.15trixterhow much inet bandwidth do you have?
21:08.19myke420247ok, it happens on most of the tollfree numbers, i'll just use a non-tollfree CID from onow on
21:08.25trixteror whatever link that does the voip traffic, since that may not be inet
21:08.27Seba_soyis a Switch of 10/100
21:08.33Seba_soyall is in local network
21:08.39Qwell[]myke420247: I've only seen that problem when calling to 8xx DIDs, so...who knows
21:08.49Qwell[]You'd think it would happen with both
21:08.54trixterso this calling card application runs solely on a local network?  how does it actually work?
21:09.00trixterhow are you testing it
21:09.05Seba_soyis an AGI
21:09.11Seba_soyasterisk only runs this AGI
21:09.31Seba_soyaudios are in GSM, but recently I pass it to g729 native.
21:10.01trixterok lets say you have a call that comes into your network ...  I understand that you said E1 somewhere and cisco sends traffic via g.729 over hte local network to the asterisk box..  how does the call get to the cisco box?  via an E1 to the pstn?  how does the call go out of your network?  via the same E1?
21:10.36Seba_soyvia E1 yes...
21:10.45trixterthere is a lot of missing information and you arent trying to provide it making this really difficult to even understand what is going on
21:10.46Seba_soycisco have 4 E1, 2 incomming, 2 outgoing
21:11.09*** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-165-64.44-151.net24.it)
21:11.10Seba_soyit is simple
21:11.14Qwell[]ditch the cisco, run the E1's right to asterisk :p
21:11.26trixterits not simple when you dodge the question and refuse to provide me with any information
21:11.28PakiPenguinhmms
21:11.31Seba_soyAsterisk does not have support for R2
21:11.34Seba_soyit sucks
21:11.35trixterremember I am not there I dont know how your network is set up
21:11.37trixterbut anyway
21:11.40*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
21:11.58Seba_soyI am searching something to do R2 signalling on asterisk
21:12.06trixteryou should have more than enough bandwidth, so after the user is authenticated does asterisk bow out?  or does asterisk keep doing stuff?
21:12.08Qwell[]Whats R2?
21:12.14trixterccitt5 r2
21:12.21Seba_soyMFC R2
21:12.25Qwell[]trixter: didn't help :p
21:12.27Seba_soylike ISDN PRI or SS7
21:12.44fugitivoyes it has
21:13.20fugitivoSeba_soy: search for chan_unicall
21:13.43Seba_soychan_unicall does not work in 1.2 asterisk for me
21:14.05fugitivoit should
21:14.10*** join/#asterisk Vco_ (i=1002@S01060010f306c408.wp.shawcable.net)
21:14.24fugitivoanyways, the support for R2 is there
21:15.36fugitivoSeba_soy: what problems you have with chan_unicall?
21:16.53sivana.
21:17.11Dr-LinuxQwell: hi there
21:17.43Dr-Linuxplease check it my test IVR >> http://pastebin.com/438127
21:18.27Dr-Linuxi can't use  "n" for net priority, and let me know if i'm missing something ?
21:19.38*** part/#asterisk hans (n=fugalh@falcon.fugal.net)
21:21.49fugitivoQwell[]: check #asterisk-dev
21:22.06Flautois there anyone using the lookup.agi came with asterisk source?
21:22.26*** join/#asterisk davro (n=davro@cpc4-ches2-3-0-cust110.lutn.cable.ntl.com)
21:24.08Qwell[]Dr-Linux: Whats is not doing?
21:24.21Qwell[]fugitivo: You sure it isn't the same bug?
21:24.46fugitivoQwell[]: this is not meetme, only iax calling through zap
21:24.57Qwell[]I don't think the other one is just meetme either.
21:24.58fugitivoand
21:25.00Qwell[]it may be though...
21:25.04fugitivomeetme works perfect too
21:25.08Qwell[]oh
21:25.12fugitivoit looks like a zaptel bug
21:25.17Qwell[]perhaps so
21:26.01Dr-LinuxQwell: sir did you check my problem related IVR ?
21:26.05Qwell[]fugitivo: What is it doing exactly?  Earlier you just said it sounds like crap
21:26.11Qwell[]Dr-Linux: yeah, whats wrong with it?
21:26.19Dr-LinuxQwell: my ivr >> http://pastebin.com/438127
21:26.33Qwell[]WHAT IS WRONG WITH IT?
21:26.45Dr-LinuxQwell: you saw there my pirorities are 1, then 2, 3 , 5
21:27.00Qwell[]okay, what about it?
21:27.03Dr-LinuxQwell: i can't use  "n"  for next pirority
21:27.07Dr-Linuxit doesn't work
21:27.18Dr-Linuxi wanna use "n"
21:27.28Qwell[]What version of asterisk?
21:27.34Dr-Linux1.0.9
21:27.40Qwell[]You can't use n with 1.0.9
21:27.59Dr-LinuxQwell: i'm creating IVR first time, so check it and suggest me if i'm wrong
21:28.07Dr-LinuxQwell: 1.2 version is fine?
21:28.16Qwell[]yes
21:28.32fugitivoQwell[]: i have a remote iax phone (atcom 320), there's only 30/40ms from asterisk it, if i call using that phone, using the x100p, the voice of the person called is perfect, but the voice of the person calling using the iax phone is robotic and sounds like crap :)
21:29.04*** join/#asterisk saftsack (n=oliver@p54A7ECD8.dip.t-dialin.net)
21:29.04fugitivoQwell[]: and, the sound of that voice, is the same i get with meetme, that's why i think it has to so maybe with zaptel as timing device?
21:29.05saftsackhi
21:29.20Dr-LinuxQwell: i can't find operator.gsm or for-operator.gsm sound file in /var/lib/asterisk/sounds/
21:29.39Dr-Linuxi'm not sure why this important file is missing or what?
21:29.55Qwell[]Dr-Linux: It might be in asterisk-sounds
21:30.02fugitivoQwell[]: and, using version 1.0.10, i have no problems (had to downgrade one hour ago to do this test)
21:30.07*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
21:30.20Qwell[]fugitivo: You're sure asterisk is using the x100p as the timing source?
21:30.26Qwell[]I don't see why it wouldn't be, but...
21:30.40fugitivoQwell[]: i don't have another timing source, how do i check that?
21:31.02Qwell[]it could just be not using a timing source?  I have no idea man...
21:31.11Qwell[]does zap show channels show a pseudo device?
21:31.15fugitivomeetme shouldn't work that way
21:31.29fugitivoyes, there's one pseudo and 1
21:31.31Qwell[]afaik, it would still "work"
21:31.46fugitivoi'm looking at 1.0.10 right now
21:31.47fugitivoholdon
21:32.04fugitivoyes, the same for 1.2
21:32.38fugitivoi don't care too much about meetme, but yes calling through zap using the iax phone
21:32.40Qwell[]not sure what to tell you
21:32.58Qwell[]x100p is flakey, heh
21:33.08fugitivobut with 1.0.10 works
21:33.13Qwell[]true
21:33.35Qwell[]fugitivo: You might want to go back to an older version of head, and see if you can find exactly when it breaks
21:34.01fugitivoi tried with a head version, and same problem, don't know which day
21:34.09Qwell[]go back a year, if it works, come forward 6 months, if it works, come forward another 3 months, then 1 month, then a week, etc
21:34.29fugitivooh, that'll be a pain in the ass with this p2 400mhz
21:34.30fugitivolol
21:34.33Qwell[]yeah :p
21:34.54Qwell[]it would narrow down exactly where the problem is though...probably :D
21:34.58trixterunf 540927 dialplan entries coming to a webserver near you!
21:36.10*** part/#asterisk Mjolinor (n=Mjolinor@cpc1-burn3-3-1-cust226.manc.cable.ntl.com)
21:36.22trixterhrm had some dupes..  its really only 523469, but its every countrys numbering plan I could find :)
21:37.17Qwell[]Do you have Qwellsakistan?  I don't follow NANPA
21:37.47trixterI would like to think that most people can find numbering plans from countries other than north america
21:38.34trixterI do see a formatting problem so I will fix that now..  also I think I am gonna take out the countrycode listings because they get in the way of this being highly useful
21:39.00*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
21:39.09*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
21:42.40LaibschIs the chan_zap.so module needed even when there is telephony hardware yet in the computer?  It appears that loading this module fails for some reason.
21:46.17*** join/#asterisk godsmoke (n=godsmoke@dsl254-096-009.nyc1.dsl.speakeasy.net)
21:46.42godsmokeI was wondering if someone here might be able to help me with some strange stuff happening with my Cisco 7960 during SIP upgrade
21:47.29*** join/#asterisk Nix (n=Nix@81.213.125.220)
21:48.01*** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net)
21:49.48Qwell[]godsmoke: ask away
21:51.35*** join/#asterisk bch (n=bch@CPE-24-26-172-197.mn.res.rr.com)
21:51.52godsmokeQwell[]: I have a 7960 I bought about a year ago -- it has some ancient firmware on it (P003AM30 according to the phone) ... this firmware isn't really mentioned much that I can find, and when I set P0S3-07-3-00 in my OS79XX.txt file, the phone requests P0S3-07-.bin from the tftp server
21:52.06Qwell[]8 char limit
21:52.10Qwell[]rename the file
21:52.25Qwell[]standard 8.3 format
21:52.35godsmokeyeah, I understand -- but, there's some mixed information about if I can upgrade directly from this old SCCP firmware to 7.3
21:52.48Qwell[]doubtful
21:53.00godsmokeCisco seems to imply it's the way to do it
21:53.02Qwell[]need to get to sccp 6.x first I think
21:53.38godsmokehttp://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2
21:53.54godsmoke3.x earlier -- does that not apply to my SCCP firmware?
21:53.55Qwell[]that the firmware upgrade matrix?
21:54.07godsmokeno, not the matrix
21:54.13godsmokeit's a few steps
21:54.36godsmokeit's titled "SCCP images 3.x/earlier and 5.x to SIP images 7.x"
21:54.54Qwell[]try just renaming the firmware file
21:54.57Qwell[]something like
21:55.15Qwell[]P0S30730
21:55.36godsmokeyeah, it just seems odd that they didn't mention any char limit
21:55.58Qwell[]they do elsewhere
21:56.23godsmokewell, rebooting -- renamed the file to what it requested last time (P0S3-07-.bin)
21:56.34Qwell[]no...that's no good
21:56.42godsmokewhy not?
21:56.48Qwell[]You won't have any way of knowing what version of firmware it's using
21:56.59Qwell[]at least with 0730 in there, it'll bw obvious...
21:57.30godsmokeso I just put P0S30730 in the OS79XX.txt file, and rename the .bin to that as well?
21:57.42Qwell[]pretty much
21:57.45bbzWould anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. --
21:57.48Qwell[]keep the .bin extension though, of course
21:58.04Qwell[]and not just the bin.  probably all of the files.
21:58.11Qwell[]perhaps make copies of them all...
21:58.20Qwell[]just in case...
22:00.11*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
22:00.12*** join/#asterisk godsmoke (n=godsmoke@dsl254-096-009.nyc1.dsl.speakeasy.net)
22:00.20godsmokehmm, xchat just died onem
22:00.48LaibschDo I need to compilee the ztdummy module to use asterisk on a computer that does not have any dedicated telephony hardware (yet)?
22:01.34bchannoying question here - how do I grab the argument passed with AGI in PHP?
22:02.45fugitivoLaibsch: you'll need it for meetme
22:03.18Laibschfugitivo: Thanks.  Good to know.  I will want to use meetme.
22:03.39trixterapp_conference doesnt need it though
22:03.57trixterbut its got some limitations such as no dtmf in conferences (ie user/admin menu self muting, etc)
22:04.12Winkietrixter: do you know much about app_conference?
22:04.24trixterbut its supposedly a faster conference application due to the way it causes transcoding to occur (I dont know if meetme has fixed some of those problems or not)
22:04.37trixterI know nothing!
22:04.42Winkiei have two asterisk servers linked with IAX, if i connect to a conference on the remote one using my local one, nobody can hear me and asterisk reports no frames in
22:04.45Winkie:(
22:04.50Winkieit's a very confusing error
22:05.26Qwell[]Winkie: Are you using iax trunking?
22:05.39trixterI dunno, can you do bidirectional stuff on other apps ?
22:05.42Qwell[]iax trunking also requires meetme
22:05.44trixterapps within asterisk that is
22:05.48trixterlike echo
22:05.59trixteriax trunking requires meetme or a timer?
22:06.06Qwell[]timer, sorry
22:06.09trixter:)
22:06.13Winkietrixter: i'm not sure, i can connect to sip phones through the IAX link
22:06.15Qwell[]You knew what I meant :p
22:06.18Winkieand that seems fine
22:06.26trixterQwell: I also knew what you said :P
22:06.29Winkiei've not tried any echo stuff
22:06.39godsmokeQwell[]: after I have 7.3 loaded, I should be seeing the Universal Application Loader on boot?
22:06.47Qwell[]godsmoke: think so, yeah
22:06.48trixterWinkie: make sure that you can call the same exact way you do the conference to echo or something and see if that works
22:06.59trixterjust to rule out something outside of app_conference
22:07.00Winkietrixter: sure, just 'Echo'?
22:07.11fugitivoNov 25 04:13:51 WARNING[5197]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 31 scheduled tasks all at once
22:07.13trixterwhy not?  if it works properly you know you can send and receive
22:07.20Winkietrixter: I mean i don't know what the command is :)
22:07.21fugitivowhat's that?
22:07.28Qwell[]Echo()
22:07.34Winkieyeah adding to dialplan now
22:09.30saftsackhi, i want to phone all my iax and sip telephone clients with one config line
22:09.42saftsackexten => 20,1,Dial(SIP/7777) thats what i have in the moment
22:09.57h3xRTFM
22:10.17saftsackive read the manual
22:10.24h3xno you didnt
22:10.30h3xcoz it tells you all about how to use the &
22:10.53saftsackh3x, i red something about sipgate
22:11.00saftsackbut what is the global entry for iax?
22:11.55Winkietrixter: nope echo recieves nothing either
22:12.20Qwell[]Winkie: Are you using iax trunking?
22:12.40trixterthen its not app_conference, or at least that isnt the first place to look
22:12.43WinkieQwell[]: I don't know, i am connecting two asterisk machines via IAX but i haven't a clue if it's 'trunking'
22:12.45Winkietrixter: indeed
22:12.50trixtermake sure your iax connection works, echo lets you do that easily, and it doesnt appear that it is
22:12.55trixter:D
22:12.55Qwell[]Winkie: Check your config.  trunking requires a timer
22:13.02trixterat least now you know a little better of where to look
22:13.11*** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net)
22:13.19WinkieQwell[]: bizarre, i can connect to sip clients fine, does it require a timer at both sides?
22:13.26Winkiesip clients on the remote asterisk, that is
22:13.27Qwell[]yes, I believe so
22:13.39Qwell[]It's probably doing a reinvite or something...I don't know
22:13.48Winkieugh, i can't add a timer in at both sides
22:13.57Winkiebecause there's no way in hell i can insert a kernel modules
22:14.00Winkiemodule either*
22:14.06Qwell[]Winkie: Then disable trunking if it's enabled.  Just check if it is
22:14.15WinkieQwell[]: how exactly do i do that?
22:14.15Qwell[]That's step 1: Check if you're using trunking
22:14.24Qwell[]check your config for the trunking option
22:14.30Winkiewell it's definately not there by default
22:14.32Qwell[]I don't know what it's called
22:15.33Winkieapparantly it's trunk=yes
22:15.34saftsackh3x,  exten => 20,1,Dial(SIP/7777) & Dial(IAX2/oliver@oliver)
22:15.36Winkieand i definately don't have that
22:15.38saftsackis that right?
22:15.43Qwell[]saftsack: no
22:15.50saftsackhow is it right?
22:15.57saftsackive no idea :(
22:16.07Qwell[]Dial(SIP/7777&IAX2/oliver@oliver)
22:16.12saftsackthank you :)
22:16.14Qwell[]That'll be $5.
22:16.18WinkieQwell[]: i'll make sure trunk=no is in my local config
22:16.19saftsacklol
22:16.20*** join/#asterisk Kuh (n=vplan@IP-213157027031.dialin.heagmedianet.de)
22:16.26fileonly $5? what a deal
22:16.41Qwell[]file: I'm like a crack dealer.  First time is always cheap
22:16.47trixterhrm I think I got all the bugs out of my automagic parser and will have that list of over 520k numbering plan entries up within minutes :)
22:17.50WinkieQwell[]: trunk is now set to no in both local and remote
22:17.55Winkiestill no frames being transmitted inward
22:18.04Qwell[]I assume you reloaded?
22:18.08Winkiei did
22:18.14Winkieiax2 reload on one, rull reload on the other
22:18.47Qwell[]well, iax2 doesn't use rtp, so..
22:18.55Qwell[]if the signalling is getting through, there is no reason for the rest
22:19.20Winkiewell i can hear the conversation, just not speak, unless it's to a sip client of the remote server
22:19.33Qwell[]speaking != conversation?
22:19.46Winkieapp_conference
22:19.48Winkiei can hear the conference :)
22:19.54Winkieincoming only
22:20.16Qwell[]sip>*>*>sip works?
22:20.37Winkieactually it's MGCP > * > * > sip
22:20.39Qwell[]might be an rtp issue afterall
22:20.39Winkiebut yes that's fine
22:20.46Winkieyou reckon?
22:20.50Qwell[]ahh...mgcp, hands off
22:20.51sm7syxHi, how do I play an announcement to the calling party ? Not the called !
22:20.56saftsackim searching for a 4 port hfc isdn card
22:21.03saftsackdo you know a good one?
22:21.19WinkieQwell[]: it's fine, the fact it's MGCP doesn't matter because it's this side of asterisk surely
22:21.32Qwell[]sm7syx: When do you want it to be played?
22:21.38Qwell[]Winkie: not if it's an rtp issue
22:21.48WinkieQwell[]: you what?
22:21.53Winkiewhy would that make any difference?
22:21.58Qwell[]it very well could
22:22.21Winkiehow?
22:22.34Qwell[]because if the rtp can't get through...
22:22.38Qwell[]how would the other end hear you?
22:22.47sm7syxQwell[], before answered... In the Dial statement
22:23.07af_where could I find some examples about call parking?
22:23.29Qwell[]af_: in the example configs, or on the wiki
22:23.32Qwell[]~wikis
22:23.35jbot[wikis] http://www.voip-info.org
22:23.36WinkieQwell[]: why would it make any difference whether it's IAX or SIP?
22:23.45Qwell[]Winkie: because iax doesn't use rtp
22:23.54WinkieQwell[]: wait i'm totally confused now
22:24.00Qwell[]lemme try to explain
22:24.12Winkieso you're saying the reason people can't hear me when i dial conf is because there could be an rtp problem excepting that iax doesn't use rtp?
22:24.18Qwell[]If *2 cannot hear the audio, but a SIP device connected to *2 can...
22:24.22*** part/#asterisk Kuh (n=vplan@IP-213157027031.dialin.heagmedianet.de)
22:24.29Dr-LinuxQwell: where can i find extra sounds in .gsm ?
22:24.41Qwell[]it could be a firewall problem, in getting the rtp from MGCP to *2.  SIP COULD be able to get the rtp, depending on network/firewall config
22:24.57Winkiei thought you just said IAX doesn't use rtp?
22:24.58Qwell[]If the call is reinvited, *2 would no longer be in the loop for audio
22:25.23Winkieright that makes sense, but my firewall is clean and prevents nothing outgoing, and the mgcp phone has canreinvite=no
22:25.23Qwell[]Winkie: If *2 is reinviting the MGCP call, it wouldn't be using iax
22:25.40Winkieit'd be using MGCP?
22:25.41Winkieor what?
22:25.49Winkiewant to see the connect string?
22:26.02*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
22:26.13Qwell[]like I said, I don't know anything about mgcp, so...
22:26.29frenzyI'm getting alot of this  Nov 25 16:43:07 WARNING[17429]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'callingcard'
22:26.56Qwell[]frenzy: so tell the user to dial something, or put a context t in
22:27.00frenzyI have to reload asterisk for it to go away
22:27.19Qwell[]erm, rule t in context callingcard
22:27.24shido6heh
22:27.27frenzyI actually get then web calling to extensions
22:27.37frenzyor receviving calls via SIP
22:27.56frenzyQwell[]: Can you give me an eg
22:27.58Qwell[]Winkie: Don't msg me...
22:28.05Qwell[]frenzy: t,1,Hangup()
22:28.22Winkiefine, jesus
22:28.26*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
22:28.32frenzyI have it set as 7777777,1,Dial(SIP/7777777)
22:28.45*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
22:28.53m160858hi
22:28.58Qwell[]I don't think I could ever do support full time...
22:28.59m160858some guy speak spanish?
22:29.04m160858i'm from peru
22:29.07Qwell[]I'd be on the news in a week or two
22:29.07frenzyQwell[]: ?
22:29.16Qwell[]"Local man kills 30"
22:29.29m160858i have problems with a broadvoice accounts
22:29.50m160858some guy wish help me?
22:29.58m160858sorry by my english
22:30.09Qwell[]m160858: Just ask your question
22:30.27frenzyQwell[]: how do I use 't' rule with dialing SIP ?
22:30.36Qwell[]frenzy: huh?
22:30.43m160858i have 5 accounts broadvoice in label register
22:30.51frenzyI dont even know what I'm talking about :P
22:30.58m160858i can call a canada & USA
22:31.01m160858but
22:31.14m160858when i want receive calls
22:31.15frenzyQwell[]: you said t,1
22:31.38frenzyQwell[]: how would I use that when calling extension 111 for sip 111
22:31.40Qwell[]frenzy: something is timing out.  Which means whatever you're dialing, isn't being matched
22:32.04m160858my asterisk no detect .. which of all my users .. send the call
22:32.36frenzyQwell[]: the damn thing started happending right after upgrading to 1.2.0
22:32.51*** join/#asterisk L|NUX (n=linux@202.141.252.82)
22:32.59Qwell[]frenzy: I'd have to see your dialplan, and an explanation of what you're trying to do
22:33.10m160858this appears when i do sip show registry
22:33.14m160858sip.broadvoice.com:5060 8182964526@s 3584 Registered
22:33.14m160858sip.broadvoice.com:5060 8182964519@s 3584 Registered
22:33.44frenzyQwell[]: simply set extension 123 to call 456
22:33.56frenzycall SIP 456
22:34.35MikeJ[Laptop]m160858, is it that the inbound call is going to the wrong ext, or are you not getting it at all?
22:35.13m160858in all
22:35.22m160858i have 10 ext
22:35.33m1608582 for each account
22:36.05Winkiefrenzy: what exactly is it you're having problems with? you're getting a timeout when dialling something?
22:36.09m160858and i to create 1 context for each 2 ext
22:36.22m160858ok?
22:36.22MikeJ[Laptop]do you see anything when you attempt to call in?
22:36.35MikeJ[Laptop]anything on the console?
22:36.38m160858but, when i call to my ext 1 or 2
22:37.02m160858receive it the ext 9 or 10
22:37.22m160858dou you understand me? .. i don't write the english
22:37.50Winkiem160858: send me your sip.conf and your extensions.conf and i'll have a look if you want, or MikeJ[Laptop] can :)
22:38.29shido6are we still on the t, problem?
22:38.35m160858where i send it ?
22:38.42Qwell[]m160858: pastebin.ca
22:38.44m160858i don't use irc normally
22:38.44Winkieshido6: i think frenzy is, but m160858 has a bunch of incoming sip problems
22:38.53Qwell[]shido6: yes...he's not given any information yet
22:38.54m160858ok, thanks
22:40.27MikeJ[Laptop]m160858, your issue is in your register statemnet
22:40.48MikeJ[Laptop]m160858, don't register to ext s, register to a specific ext #
22:40.56m160858i do
22:41.19MikeJ[Laptop]paste one of your register lines from sip.conf (minus any passwords(
22:41.22*** join/#asterisk benjk (n=benjk@f8a01-0357.din.or.jp)
22:41.34Qwell[]one of the ones that aren't working
22:41.56fileenough messing with app_timeout_supreme for now
22:42.07MikeJ[Laptop]app_superfile!
22:42.12filemeep!
22:42.23Qwell[]mmm, taco_supreme
22:42.24MikeJ[Laptop]someone please save me from vbscript
22:42.31fileI found a small mistake in app_skel.c ;)
22:42.38MikeJ[Laptop]ewww
22:42.39MikeJ[Laptop]worse
22:42.42filea LOCAL_USER_REMOVE before LOCAL_USER_ADD has been done
22:42.44m160858just done but all every thinks
22:42.53Winkiefile: don't you mean app_timeout_awesome? >:(
22:43.03filenope - supreme!
22:43.15MikeJ[Laptop]m160858, can you please paste me one of your broken register lines from sip.conf
22:43.17fileit's almost 400 lines long so far
22:43.24fugitivoapp_timeout_uber_elite
22:43.41Qwell[]fugitivo: That'll be a fork later
22:43.46fugitivohehe
22:43.48m160858are going to seem
22:43.50benjkyou are working on call timeout?
22:43.57fileyes
22:44.07benjkcan I make a suggestion?
22:44.13filemaybe
22:44.27MikeJ[Laptop]m160858, puede usted pegarme por favor una de sus líneas quebradas del registro de sip.conf
22:44.39fugitivolol
22:44.40MikeJ[Laptop]yay babel fish
22:44.41fugitivo"quebradas"
22:44.48benjkwewant to set up a community VOIP service with free PSTN calls
22:44.51fugitivom160858: do you speak spanish?
22:44.52MikeJ[Laptop]hell.. I don't speak spanish..
22:45.06filespanglish!
22:45.08m160858this is my extensions.conf
22:45.09m160858http://pastebin.ca/31247
22:45.10fugitivom160858: hablas español?
22:45.22MikeJ[Laptop]not extensions, sip.conf
22:45.24benjkone of the transpiring ideas is to introduce a waiting period of -say- 5 minutes if a maximum call duration was exceeded
22:45.33Qwell[]eeps, default
22:45.55filebenjk: and call timeouts would have what to do with that?
22:45.56*** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner)
22:46.08m160858this is my sip.conf
22:46.09m160858http://pastebin.ca/31248
22:46.18m160858si hablo español
22:46.22m160858gracias gracias
22:46.23m160858jeje
22:46.25benjkwe are talking maximum timeout right?
22:46.30m160858entiendes mi problema?
22:46.39fugitivom160858: cual es el problema?
22:46.46*** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
22:46.49MikeJ[Laptop]m160858, first off... you DO NOT want your outgoing extensions in your default context.
22:46.55filewell that already exists, I'm writing a module to do group timeouts so if a person called twice, each timeout would be adjusted because two calls are active
22:47.02m160858how i do this?
22:47.06MikeJ[Laptop]it could allow people to use your pbx to make outbound calls
22:47.22m160858do you have some example?
22:47.26Qwell[]that sip.conf is...yeah
22:47.30benjkthe waiting period for a follow on call I am talking about is rather similar
22:47.33*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
22:47.54filebenjk: that shouldn't be done in the core though... and I wouldn't do that in this application as it's stateless once a group is empty
22:48.05benjkI see
22:48.07MikeJ[Laptop]m160858, in the register statements in sip.conf add \123456 if 123456 is the extension you want it to go to
22:48.13MikeJ[Laptop]to each one
22:48.13m160858I am new in this
22:48.39Qwell[]What is that in the first colon for?
22:48.41MikeJ[Laptop]ouch
22:48.50MikeJ[Laptop]m160858, does that make sense
22:48.53benjkany ideas how to best achieve th waiting period for follow on calls?
22:48.59m160858i think was /12345
22:49.00benjkapproach wise I mean
22:49.19m160858i agree this, but it's the same
22:49.34MikeJ[Laptop]wait.. those registers look not right
22:49.34filebenjk: well you can have the h extension be executed when it's hung up... and go to an AGI... then when they try to call again get the AGI to check the presence and timestamp...
22:49.47fileif it's 5 minutes or less, play a message... if it's over delete the file
22:49.56fileor a database entry
22:50.01Qwell[]MikeJ[Laptop]: yeah, thats what I said :p
22:50.02kippihi
22:50.08kippiI am getting this errir
22:50.09kippiOuch ... error while writing audio data: : Broken pip        e
22:50.13filekippi: that's normal
22:50.17Qwell[]Broken pip?!
22:50.17Qwell[]e
22:50.24benjkyeah I was thinking about a database entry
22:50.37kippihow can i fix it?
22:50.41MikeJ[Laptop]Qwell, well I had to actually look
22:50.49kippias it is stoping * from loading
22:50.50filekippi: it's not a problem you can fix, as it's not a problem
22:50.52MikeJ[Laptop]m160858, your register statements are wrong
22:50.52Qwell[]MikeJ[Laptop]: I was at a loss for words
22:50.55filekippi: the problem is elsewhere
22:51.11MikeJ[Laptop]m160858, register => PHONENUMBER:PASSWORD@sip.broadvoice.com
22:51.14MikeJ[Laptop]umm
22:51.21kippifile: how can I find out where the error is?
22:51.26MikeJ[Laptop]m160858, register => PHONENUMBER:PASSWORD@sip.broadvoice.com/EXTENSION
22:51.28filekippi: look on the screen and see before it?
22:51.31Qwell[]Does the /exten always work?
22:51.35Qwell[]Are there instances where it might not?
22:51.39m160858yes, i do it ... this
22:51.46fileQwell[]: depends on the provider
22:51.46MikeJ[Laptop]m160858, everything in all CAPITALS should be replaced with your settings
22:51.49m160858already do it
22:51.58kippifile: Parsing '/etc/asterisk/zapata.conf': Found
22:52.08Qwell[]m160858: No, you're doing it much differently
22:52.11MikeJ[Laptop]m160858, paste what you have now for sip.conf
22:54.08kippiis it to do with mpg123?
22:54.26filekippi: I doubt it, pastebin a full log of what you see when asterisk starts
22:54.33kippiok
22:55.14fileMikeJ[Laptop]: it's the eye of the code! O.O
22:55.38kippimy pastebin is http://pastebin.com/438214
22:55.53filechan_zap is thy problem
22:56.00filesomething is making it go insane
22:56.49kippifile: how can I track it down?
22:57.50kippi:(
22:58.33*** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar)
22:58.48kippican anyone else help me with this error?
22:58.57LaibschWhere can I get a list of the internal telephone nummbers like *99 and *77?
22:59.31fileLaibsch: for what? asterisk doesn't have any by default
22:59.32*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
23:00.13Laibschfile: Thanks you for the reply.  They are in extensions.conf.  I used that had been prepared by somebody else.
23:02.38Qwell[]aka AMP
23:03.03fileunclean!
23:03.15fugitivolol
23:04.23*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
23:05.37Nuggethttp://es.gnu.org/~jemarch/images/rms_clip.jpg
23:05.56fugitivohehe
23:06.31Qwell[]nice
23:08.36fileI've got chills.... they're multiplying!
23:08.39fileand I'm losing control
23:08.46Dr-LinuxQwell: whats disfference between application >> playback and background  in ivr?
23:09.01fugitivobackground waits for dtmf tones
23:09.05Qwell[]show application background
23:09.07Qwell[]show application playback
23:09.14Qwell[]and stop asking me every damn question...kthx
23:09.23Dr-Linuxok :)
23:09.28fugitivoQwell[]: why my cat is pregnant?
23:09.41fileQwell[]: why do I see alive people?
23:09.43Qwell[]fugitivo: because she's a little whore
23:09.50Qwell[]file: see above
23:10.00fugitivoQwell[]: why my dog eats his own poop?
23:10.08filewell this is interesting, I wrote "hrm a combo" in a notepad
23:10.11Qwell[]You would too if you were a dog
23:10.24MikeJ[Laptop]Dr-Linux... nothing.. if you use the super duper secret option for background
23:10.41filewhich turns it into... playback!
23:11.01MikeJ[Laptop]yep
23:11.13Qwell[]which super secret option?
23:11.27fileit's secret
23:11.30MikeJ[Laptop]pbx.c   read the source
23:11.31fileonly the cool people know it
23:11.49filehint: what does playback start with?
23:12.00Qwell[]p?!
23:12.02sbingnerplay
23:12.02filehaha I just noticed something...
23:12.04sbingner:b
23:12.10MikeJ[Laptop]?
23:12.21fileMikeJ[Laptop]: if you look at the options declaration it spells SNMP
23:12.25fugitivoAST_APP_OPTION('p', BACKGROUND_PLAYBACK),
23:12.36MikeJ[Laptop]heheh
23:12.41MikeJ[Laptop]that was unitentional
23:12.47filea likely story
23:13.05*** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl)
23:13.05MikeJ[Laptop]if I meant to do it.. it would have been somthing much more amusing than that
23:13.19*** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3)
23:13.25*** part/#asterisk Lurr (n=pr0ph3t@63.69.20.3)
23:13.30fugitivowhat does 'n' do? noanswer??
23:13.30*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
23:13.56MikeJ[Laptop]on playback?
23:14.00fugitivobackground
23:14.01MikeJ[Laptop]or background
23:14.13MikeJ[Laptop]yes, for that, you would type show application background
23:14.19fileit's an evil way to be sneaky
23:15.01MikeJ[Laptop]I like nice ways to be sneaky
23:15.07fugitivodon't answer before playing the files, what's the use of that?
23:15.17fileearly media
23:15.34fileI won't say more as it's sneaky what you can do
23:16.18fugitivo:/
23:16.58Qwell[]free pr0n audio
23:17.25fileQwell[]: go back to work, slacker!
23:17.33Qwell[]I am working
23:17.43fileliar
23:18.05Qwell[]Shouldn't you be working?
23:18.10filenope
23:18.23Qwell[]likely story
23:18.30fileyup, VERY likely
23:20.27Qwell[]god chatzilla sucks
23:20.33Qwell[]/leave doesn't even work properly
23:20.49*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
23:20.53Qwell[]So, even though I don't have the window open, I'm still in #asterlink for some reason.
23:21.11filenice!
23:21.19Qwell[]file: So, just pretend I'm not there...so you can't poke me back! :p
23:21.23fileokay
23:22.05Qwell[]was supposed to be a slick driveby poking
23:22.18filewell you failed horribly
23:22.23Qwell[]chatzilla failed
23:22.55Dr-LinuxQwell: i have 4 port FXO card, and currently i have 1 PSTN line, so what you recommend on which port should i use the line for now?
23:23.07Qwell[]Dr-Linux: What did I say earlier?
23:23.11Qwell[]Don't direct questions at me
23:23.17filebrb
23:23.19Dr-Linuxok sorry
23:23.35Qwell[]use whichever port you like
23:23.35Dr-Linuxi have 4 port FXO card, and currently i have 1 PSTN line, so what you recommend on which port should i use the line for now?
23:23.56Dr-LinuxThanks
23:24.40*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
23:24.55file[laptop]yays
23:24.58file[laptop]here I am
23:25.46moraleanyone use asterisk in their home with a tdm400p card and analog phones? im trying to get my extensions.conf working all spiffy and would like to see some examples/ideas..
23:26.11benjkwhat do you mean by 'spiffy' ?
23:26.49fugitivomorale: yes, what do you mean by spiffy?
23:27.28file[laptop]oyo como va!
23:27.44benjkfor extensions.conf it doesn't really matter whether or not its a tdm400
23:27.55moralespiffy? i mean working better than i have it now :)
23:28.24benjkthat's a little too imprecise to help you though
23:28.27fugitivomorale: well, we don't know what it's working now
23:29.18trixterhttp://www.0xdecafbad.com/uploads/bigcountry.csv.gz - its uploaded.  3.5MB compressed 43MB uncompressed 579,887 entries, every numbering plan I could find with country, provider, whether its geographic (and where), mobile, premium, special, short codes, etc..  prolly the most complete list out there that is free
23:29.44benjktrixter, you are still here?
23:30.27*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
23:30.34trixternope
23:30.39trixterI left days ago
23:31.13benjkwell, I just had a good night's sleep and you are still here ;-)
23:31.24*** join/#asterisk digime (i=digime@219.92.173.11)
23:32.06moralefugitivo: not very much at all, i'd like to get it to ring my phone, and provide voicemail with db storage or file system storage..
23:32.07benjktrixter, you should add 1 more entry though
23:33.15Qwell[]buahahaha!
23:33.19Qwell[]file[laptop]: I did it
23:33.29file[laptop]oh noes!
23:33.35Qwell[]took a /quote part
23:33.44Qwell[]god that was difficult
23:33.54file[laptop]I have sekret and magikal powers
23:34.09trixterwhich entry?
23:34.20benjktrixter: doesn't matter
23:34.34trixterwhat a watermark entry?
23:34.37benjktrixter: if you add 1 more entry you get a lucky number
23:34.49trixterahh
23:34.49benjk579,888 entries
23:35.02benjkor remove 999
23:35.08trixterheh
23:35.15benjkthen you get 578.888
23:35.21*** join/#asterisk nagl (n=nagl@213.235.241.6)
23:36.24krischnatrixter: download link does not work for me
23:37.15trixtereeps! I just sent that to the mailing list ...  lemme check
23:41.32*** join/#asterisk Rav1974 (n=r@pool-68-161-69-3.ny325.east.verizon.net)
23:43.32*** part/#asterisk davro (n=davro@cpc4-ches2-3-0-cust110.lutn.cable.ntl.com)
23:46.22*** part/#asterisk Laibsch (n=Laibsch@p54B99A0F.dip0.t-ipconnect.de)
23:52.53Rawplayerwhere can i put in xlite that it should use sip?
23:53.04Qwell[]xlite always uses sip
23:53.05fugitivoxlite uses only sip
23:53.37RawplayerNov 25 23:38:36 NOTICE[18903] pbx.c: Cannot find extension context 'sip'
23:53.41Rawplayerwhat does that mean?
23:53.46fugitivodamn, my /usr is full
23:54.26Rawplayerfugitivo: delete kernel source
23:54.54fugitivogood idea, i'm going to remove 5 older versions
23:55.21Qwell[]old kernels modules and such take up a bit of room too
23:55.43dudesRawplayer - it probably means you just have sip and not sip/context/${EXTEN}
23:56.28*** join/#asterisk DrJES (n=macleajb@blk-222-132-53.eastlink.ca)
23:58.04*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
23:59.00*** join/#asterisk [TK]D-Fender (n=aoulton@66.11.164.239)
23:59.47trixterkrischna: the download should work now :)
23:59.53DrJESWould anyone suggest why I can not get 2 TDM400Ps to work in one PC? Each one works separately and ztcfg -vv shows all 8 channels are ready to go, but asterisk is unable to configure the new ports (5.6.7.8) ?

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