irclog2html for #asterisk on 20051210

00:00.07QwellI need to find a protocol to use between my * boxes
00:00.10Drukenangler: i'd be lookin for a german voip provider...
00:00.23Qwelliax2 is the obvious, sip is good, but...I want obscure
00:00.40Qwellsccp can't do server to server...I don't think mgcp can either.  maybe h323?
00:00.46[hC]haha
00:00.47*** join/#asterisk lodeon (n=not4u@h75n5c1o1023.bredband.skanova.com)
00:00.49twisted[asteria]lol
00:00.51[hC]I just want something thats reliable
00:00.53twisted[asteria]sip dude
00:01.04[hC]sup twisted
00:01.10DrukenQwell: write your own ??
00:01.24QwellDruken: chan_qwellhack?
00:01.34Qwellor, as it's known internally, chan_crap
00:01.40Drukenexactly
00:01.45Drukenchan_crap
00:01.46Druken:)
00:01.51twisted[asteria]chan_carrierpigeon
00:01.57Qwellit's been done
00:02.06Drukenowls?
00:02.12twisted[asteria]no way man
00:02.15QwellI need something obscure
00:02.15twisted[asteria]asterisk prints punchcards
00:02.29twisted[asteria]the carrier pigeons carry the punch cards to another site and drop them into the asterisk server there
00:02.37DrukenQwell: analog :)
00:02.39QwellWhat if the boxes are right next to each other?
00:02.45Drukenthat obscure enough for ya?
00:02.47twisted[asteria]Qwell, TDMoE
00:02.47QwellDruken: too easy
00:02.57Qwelltwisted[asteria]: Now you're thinking
00:03.29twisted[asteria]i'm always thinking.
00:03.30Qwells/think/talk/
00:03.41Drukenmorsecode
00:03.51Qwellchan_braile?
00:03.54twisted[asteria]well
00:04.06twisted[asteria]you could do TDMoFR
00:04.12twisted[asteria]if you wanna get obscure
00:04.17[hC]Mmmm TDMoE.. I wanna do tdmoe
00:04.21QwellFR?
00:04.24twisted[asteria]frame relay
00:04.28[hC]for redundancy
00:04.30shido6...
00:04.31Qwellright...yeah, I don't think so
00:04.37Drukeni that would be kinda neat, if someone had the morsecode in audio format, and you could use it in an app like saynumber()
00:04.52QwellDruken: Doesn't seem like a very hard app to write, really
00:04.56[hC]that should be a dtmf option... inband, rfc2833, morse
00:05.02twisted[asteria]lol
00:05.07twisted[asteria]pulse.
00:05.10twisted[asteria]same difference
00:05.20[hC]cause existing dtmf doesnt have enough sensitivity, throw some dot dashes in there...
00:05.21[hC]haha
00:05.22twisted[asteria]one is voltage one is audio
00:05.29DrukenQwell: probably not...
00:05.34twisted[asteria][hC], well, you DO have A B C and D
00:05.37twisted[asteria]not to mention # and *
00:05.37Qwellmorse numbers are the same as pulse?
00:05.45[hC]oooo....
00:05.52Drukeni just figured alot of dickhead hams could have fun with something like that
00:06.01twisted[asteria]morse numbers i thought were just . .. ... .... ..... etc..
00:06.11Qwelldunno, probably are
00:06.49*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
00:06.52Qwellhell, you could probably just do morse with playtones or similar
00:06.55*** join/#asterisk _DAW (n=bob@adsl-222-51-184.msy.bellsouth.net)
00:07.01Qwellwhatever core funcs it calls
00:07.51twisted[asteria]..-./..-/-.-./-.-//-.--/---/..-//--.-/.--/./.-../.-..
00:08.17fugitivo... - - - ... (sos)
00:08.24_Sam--hey i use the GoToIfTime command...GotoIfTime(11:00-19:00   but sometimes when its after 19:00 it still goesto
00:08.28Drukenoh shit....
00:08.35Drukenfugitivo: did you actually look that up?
00:08.39_Sam--until like 19:01 or sometimes later
00:08.54fugitivoDruken: no, i knew it
00:09.03Drukenyou a ham ?
00:09.07fugitivoeverybody should know how to do a SOS in morse code
00:09.15fugitivono, that's the only thing i know :)
00:09.19twisted[asteria]--/---/.-./..././/-.-./---/-.././/../...//./.-/.../-.--//--./..-/-.--/.../.-.-.-/.-.-.-/.-.-.-//-.-./---/--/.//---/-./.-.-.-
00:09.27Druken:)
00:09.35Drukeni was learning morse at one point
00:09.53fugitivotwisted[asteria]: the same to you!
00:09.57twisted[asteria](yes, i'm actually saying things)
00:10.05QwellCan morse to cyrillic?
00:10.06fugitivomorse code is too geek
00:10.07Qwelldo*
00:10.25The-Darkbump --- quit --- no help
00:10.31twisted[asteria]and FYI:  /'s are letter separators, and //'s are word separators
00:11.12fugitivotwisted[asteria]: do you usually chat in morse code?
00:12.10Drukenangler: find one?
00:13.02anglerDruken, looking some... maybe i can get one of our resellers to hook me up :)
00:13.25twisted[asteria]fugitivo, no
00:14.50*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
00:15.17Drukenangler: ok :)
00:15.32harryvvBTW looking for a UPS for my asterisk box what is a good one that would possibly offer exernal battery hookup?
00:15.50DrukenAPC :)
00:16.12Drukenbut apc isn't cheap
00:17.06harryvvI know
00:17.33*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
00:18.42*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
00:18.57redder86who is responsible for the new (IAX) jitterbuffer in 1.2
00:18.58redder86?
00:19.05redder86stevek?
00:19.54Drukeni must say i'm almost afraid to upgrade... all my shit works right now...
00:19.58Drukenmay not if i upgrade
00:19.59Drukenhehehe
00:20.53*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
00:21.51*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
00:23.23jarrodmust always have an  identical server to upgrade to
00:23.25jarrodthen if it works
00:23.31jarrodleave it
00:25.55*** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com)
00:27.27*** join/#asterisk anthm (n=anthm@h46088983.area4.spcsdns.net)
00:27.28*** mode/#asterisk [+o anthm] by ChanServ
00:29.19asterboyAnyone have a detailed list of changes between 1.2.0 and 1.2.1???
00:30.02asterboyThe ChangeLog is hard to digest.
00:30.03zigmanchangelog
00:33.55harryvvWhat does it take to make the second line on my ip500 light up when a call is comming in?
00:34.30robl^harryvv: a hint!   :)
00:35.06*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
00:35.33harryvvyea never really tested my system for this.
00:35.49harryvvBut of course, I just got my 1877 DID and that works.
00:35.52shido6you have to set the second line to something else
00:36.05robl^harryvv: are you talking about lights that show the status of other SIP accounts/extensions?
00:36.08shido6register the second line with the username/pass you want to ring
00:36.27harryvvrobl just a expression. light up meaning im getting a call on the second line.
00:36.50harryvvshidow in the phones cfg files?
00:37.08robl^harryvv:  ohhh!  just register it as a seperate SIP account in /etc/asterisk/sip.conf
00:37.16harryvvokay
00:37.46robl^then tell the phone that line2 has a user/password/server..  just like you did for line 1
00:38.18bsdfreakheh
00:38.19ManxPowerhardwire, Register EACH line as a seperate sip.conf entry (we use the MAC address, all lower case, with a -a -b -c -d, etc suffix for each line)
00:38.26bsdfreakhas anyone upgraded from 1.x to 1.2.1 recently?
00:38.45ManxPowerThen set the phone to only accept 1 call per line (it's a config option new to 1.6.x I think)
00:38.48*** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca)
00:39.30hardwirehey..
00:39.38*** join/#asterisk marc32344 (n=marc3234@206-248-152-83.dsl.teksavvy.com)
00:39.55bsdfreakanyone upgraded their sip firmware from 2.x to 3.x?
00:40.05robl^ManxPower is the Polycom god! :)
00:40.38harryvvmanx, then anytime i ask a question you should know it :)
00:40.46ManxPowerI use the not-latest version of the Polycom BootROM (since I have a mix of x00 and x01 phones, with the 1.6.2 SIP firmware
00:40.56marc32344what happens if the ip address of my asterisk changes. Do all sip clients need to get an update?
00:41.13shido6http://pastebin.ca/33144
00:41.15ManxPowerharryvv, I only admit to using polycoms when I have the time to actually help someone.  It's not simple.
00:41.28shido6harryvv http://pastebin.ca/33144
00:41.29bsdfreakheh
00:41.37harryvvmanx, i know thay are a pain
00:41.44harryvvbut i got mine up and running
00:41.49harryvvworks almost flawlessly
00:41.58shido610.0.0.6 being the * box
00:42.21bsdfreakok
00:42.30bsdfreakCAN you upgrade a 2.x hardware sipura to 3.x firmware?
00:43.26bsdfreakim guessing not
00:44.05harryvvshidow6 thanks for that.
00:44.11harryvvThat information
00:46.58*** join/#asterisk Craziman2 (n=Craziman@63.108.128.250)
00:48.47ManxPowerharryvv, Oh, they are WONDERFUL phones, but the learning curve is pretty steep for a newbie.
00:49.41ManxPowermarc32344, Make sure the IP address of your Asterisk server does NOT change.
00:50.04ManxPowerIf it does change you will be glad the phones are configured to use DNS hostname and not an IP address.
00:50.19moraleafter i upgraded to asterisk 1.2.1 i get the error: WARNING[26464]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '101', when i do a show voicemail users it does show that extension 101 is defined.
00:50.51*** join/#asterisk kfuq (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
00:55.18shido6screw that
00:55.22shido6whats voicemail.conf say?
00:55.25shido6pastebin.ca
00:55.55moralettp://pastebin.com/457892
00:56.57marc32344i have 4 servers running *, what switch should I get?
00:57.23moralewhat switch?
00:58.40marc32344for connecting the servers together
01:00.16moraleit doesn't really matter.. if you got the money get a managed switch
01:01.24asterboysure would be nice if they could add a clearer change list from 1.2.0 to 1.2.1
01:01.53asterboyI'll have to digest the ChangeLog...yuk
01:01.56[hC]argh... i have some sip devices that are constantly re-registering. linksys pap2's
01:02.56asterboy[hC]: is that a cause of the pap2's or is that what your sip devices are re-registering as?
01:03.28[hC]asterboy: its either the pap2 doing it, or the internet connection that those people are on. not sure.
01:03.34[hC]playing around with expiry times now
01:03.52asterboy[hC]: but you are using pap2s?
01:04.00*** join/#asterisk javar (n=javar@69.79.133.185)
01:04.12[hC]asterboy: yeah pap2-na's
01:04.47SkramXSee all yall, cats.
01:05.23asterboy[hC]: can you pastebin your sip.conf?
01:05.59[hC]sure.
01:06.21[hC]one thing to note, i just changed defaultexpiry and maxexpiry from both 45 to the values they are currently, to see if it makes a difference...
01:06.41asterboy[hC]: ok....Also, your http stup screen.
01:06.47asterboy[hC]: setup
01:07.07asterboy[hC]: you configuring those pap2's via http right?
01:07.26[hC]yep. i dont have access to them at this moment, but yes i am
01:07.30[hC]http://pastebin.ca/33146
01:07.31[hC]there
01:07.41[hC]the globals and the peer in question
01:07.53*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
01:09.33asterboy[hC]: how long betweeb re-registration attempts?
01:09.41[hC]not sure...
01:09.51[hC]sometimes hours, sometimes 20-30 minutes? I dont really pay attention
01:10.01[hC]i just see registration messages from time to time
01:10.42asterboy[hC]: did you turn on sip debug?
01:10.45trixterI have a 50% discount code for etel which I heard can be used with a 10% off or something in addition if bought before Jan 9 or something..  if anyone is interested
01:11.29[hC]asterboy: nope not yet. i turned down the expiry time cause they kept losing registration, i think because of nat timeouts on the router they use at that location.. not sure..
01:11.35[hC]havent really dug into it too deep yet
01:12.30asterboy[hC]: are those "unlocked" paps?
01:12.41asterboy[hC]: must be if your using them on your own system
01:14.17shido6I have unlocked paps
01:14.44asterboy[hC]: shido6, are you getting sip re-registrations?
01:15.29[hC]astterboy: yeah, the -na's are unlocked. are you having the same problem??
01:15.53asterboy[hC]: not I, but wondering if shido6 is.
01:16.17asterboy[hC]: I'm bidding on some Vonage locked ones on ebay...so I will be going down this path.
01:17.50*** join/#asterisk Inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
01:18.24asterboy[hC]: woowoo?
01:18.28asterboy[hC]: :P
01:19.41asterboy[hC]: sip.conf looks good.
01:19.58*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
01:20.06shido6how many pap2-na's do you need?
01:20.52asterboy[hC]: Just 1 for now to play with...then I want to start purchasing them for customers I sell my telephone service to.
01:21.12shido6switch to advanced view click on SIP
01:21.25shido6what does reg max expires say
01:21.34shido6and invite expires
01:21.48shido6there's a few numbers there - what are they?
01:22.10asterboyYa, I requested the http page for pastebin.
01:22.18shido6$63 new
01:22.24*** part/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl)
01:22.44*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:23.01*** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net)
01:24.32*** part/#asterisk m160858 (n=jsaenz@200.89.12.46)
01:24.40[hC]asterboy: i changed that on purpose :)
01:25.13asterboypastebin?
01:26.15moraleargh, this voicemail stuff is all broken in 1.2.1
01:26.43[hC]what stuff? Im just about to upgrade
01:26.50moralei can't find voicemail boxes anymore
01:26.59Nuggethttp://news.netcraft.com/archives/2005/12/09/critical_security_hole_in_phpmyadmin.html
01:27.03Nuggetgo go gadget mysql!
01:27.43moraleanyone use realtime voicemail?
01:28.03[hC]ahh.. realtime.
01:28.08[hC]i feel better then.
01:28.09[hC]:P
01:30.25bsdfreakheh
01:30.41bsdfreakanyone upgraded their sipura to 2.0.13 and have it stop properly registering?
01:30.52bsdfreakin that asterisk now comes back with "proxy authentication required"
01:31.00bsdfreakeven though the sipura reports its successfully registered
01:31.07bsdfreakinbound calls work, but outbound do not
01:31.15bsdfreak(come back with a BUSY)
01:31.40*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
01:31.47Flautobsdfreak, 2.0.13 has been very reliable to me
01:31.54bsdfreakim guessing i should reset to factory defaults and reenter my config
01:31.59bsdfreakflauto: hmm..
01:32.02redder86stevek: around?
01:32.04bsdfreakwell as soon as i upgraded a few things broke
01:32.08Flautobsd, what service you are using
01:32.15Flautoor your asterisk?
01:32.21bsdfreaksipura 3000 - > asterisk 1.0.9 -> asterlink
01:33.03*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
01:33.31bsdfreakand when i show the sip peers its there
01:34.50morale[hc] i wasn't using realtime but now i am at it works
01:35.30Flautothen, it should be registered
01:35.39Flautoi have two spa3k
01:35.40bsdfreakbut it does not show it in the sip show registry
01:35.48bsdfreakso it isn't registered
01:35.49bsdfreak:|
01:35.56Flautoone is using 2.0.13 and other one is on 3.0.17
01:36.03bsdfreakyeah
01:36.12bsdfreakso one has 2.0 hardware and one has 3.0 hardware?
01:36.25Flautomine dont' show in sip show registry at all
01:36.30Flautobut they are registered
01:36.31bsdfreakah interesting
01:36.36bsdfreakwell * comes back with
01:36.43bsdfreak"Error 407: Proxy Authentication Required"
01:36.48bsdfreakevery time i try to call out
01:36.55bsdfreakbut incoming calls work to the extension
01:37.03bsdfreakso i guess it is registered, but something is messed up heh
01:37.24*** join/#asterisk TedC (n=ted@gray.impulse.net)
01:37.32bsdfreakalso calls to the pstn dont work
01:37.42bsdfreakthey'd fwd'd to asterisk (so i guess that's sort of an outbound too)
01:37.48bsdfreakgive the same proxy authentication required error
01:37.48bsdfreakheh
01:37.59bsdfreakall i did was upgrade from 2.0.11 to 2.0.13
01:39.02Flautothat is strange
01:40.33*** join/#asterisk lodeon (n=not4u@h75n5c1o1023.bredband.skanova.com)
01:41.48moralesweet it works now
01:42.14moraleexcept it doesn't change my voicemail password in my database
01:44.35*** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com)
01:46.23*** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk)
01:46.26Flautobsd, would you want show your sip.conf and the config of your spa
01:55.24*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
01:57.32MstlyHrmlsManxPower: why would having an x00 stop you from upgrading BootROMs? It worked OK for me on a 500...
02:02.25*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
02:04.41*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-83.cybersurf.com)
02:06.43*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
02:10.27moralew
02:10.35moralehere can i find docs on voicemail+realtime?
02:10.44Qwellwiki
02:10.52moralethats all for 1.0.9 though
02:11.04Qwellno it isn't
02:11.22robl^its all for stable.. :)
02:11.30robl^stable is 1.2.1
02:11.59JunK-Yyay, mark and asterisk, cover of von mag
02:12.17robl^is he in his hot tub? :)
02:12.31JunK-Yhehehe
02:16.08robl^is there a way to get a "printer friendly" view from the voip-info.org wiki?  I could have sworn there used to be a printer icon/link on each page
02:16.23Qwellrobl^: They changed wikis
02:16.31robl^ohhhh
02:16.38Qwellfew months ago I think
02:16.40Qwellmaybe 2-3?
02:16.57tzangeryikes
02:16.59robl^I noticed it was a little diggerent
02:17.04robl^different
02:17.10Dr-Linuxhi robl^
02:17.13tzangerI'm trying to fix my snowblower carb
02:17.32robl^hi Dr-Linux
02:17.33tzangerthis is complex
02:19.18Dr-Linuxhttp://pastebin.com/457960  << this is my extensions.conf and queues.conf, my queue is playing fine, when i hit 4444 extension, it tries mentioned member extension in queues.conf, sor for the testing purpose, i put only mine extension in queues.conf, but what i suppose to do?
02:19.34Dr-Linuxits not directing call to me
02:20.12Dr-Linuxkindly help me if i'm missing something.
02:20.23Qwell"extensions.conf" ?
02:20.31Qwellerm, where'd that little circle go?
02:21.08Dr-LinuxQwell: extensions.conf is in pastebin link that i given above
02:21.18Qwellyeah, but it was funkified on the cat
02:21.25Dr-LinuxQwell: goes no where same things again and again
02:21.37Dr-Linuxgreetings, music on hold, then wait then same same
02:21.44QwellSo, while the user is waiting in the queue, they hit 4444?
02:22.38Dr-LinuxQwell: while user is listing music and wating in the queue, i'm not sure what should be here, and where should he/she go ..
02:22.48Qwellit should ring one of the agents
02:23.15Dr-LinuxQwell: you mean, the member thats exist in queues.conf ?
02:23.17Dr-Linuxright
02:23.18QwellIs 4092 defined in sip.conf?
02:23.24Dr-Linuxyes
02:23.33Dr-LinuxQwell: 4092 is my own extension
02:23.36Qwellokay, and what does the CLI say?
02:23.42Dr-Linuxi'm using X-Lite, so i put that just for checking
02:23.59Dr-LinuxCLI says nothing .. just greeting messages
02:24.20Qwellverbose is high?
02:24.27QwellIt should tell you where the call is going
02:24.53Dr-Linuxbut if i uncomment all the extensions, CLI hit only three extensions 2008, 2007 and 2006 and says host not found ..  somewhat
02:25.26file[laptop]haha
02:25.32Qwellfile[laptop]: ?
02:25.45*** join/#asterisk xtr (i=01928375@S01060012174cc0e1.vf.shawcable.net)
02:25.46*** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it)
02:25.48Dr-Linuxso for the testing purpose, i commented all the extensions, and only put there mine .. but i doesn't show mine or anything :S :(
02:26.06*** join/#asterisk Majestik (n=MajestiK@S0106000024c058cc.ed.shawcable.net)
02:26.09Dr-Linuxi'm not sure what i'm missing :S
02:26.21*** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com)
02:26.22file[laptop]Qwell: I was mildly amused by the few people's departures
02:26.28Qwellahh
02:28.12file[laptop]I gave the support agents my e-mail.. what was I on when I did that
02:28.51Qwellmv /home/jcolp /home/jcolp-hiding
02:29.05file[laptop]indeed
02:29.45[hC]huh.. it appears that after i downgraded to 1.2.1 from cvs head, chanspy makes * segfault
02:30.11file[laptop]Nugget !!!
02:30.16Nuggetmoo
02:30.18file[laptop]Nugget: do you come with fries?
02:30.24Nuggetsuper size!
02:30.27file[laptop]yay
02:30.34Nuggethttp://natpool.slacker.com:2000/live.html?preset=dialup  <-- useless, but kinda neat
02:30.47file[laptop]Nugget: also won't open for me
02:30.51Nuggetbummer
02:30.56Dr-LinuxQwell:
02:30.56file[laptop]so very useless
02:31.05Qwellgrr
02:31.10Qwelldon't msg me
02:31.17Dr-Linuxthese are the CLI messages
02:31.17Qwellunless you owe me money
02:31.19Nuggetoh, oops.  :)
02:31.22Qwell!pb
02:31.24Dr-Linuxok
02:31.26Qwell~b
02:31.27jbotpicobot: c
02:31.27Qwellbah
02:31.32Nuggettry now
02:31.33Dr-Linux:S
02:31.39NuggetI forgot to refresh the mDNS port forward
02:31.46*** join/#asterisk Inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
02:32.37QwellNugget: wow, that's kinda cool
02:32.54*** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com)
02:33.42Nuggetyeah, it is.
02:34.01Qwellneeds to be higher res, heh
02:34.01Dr-LinuxQwell: well its working now, :)
02:34.25Dr-Linuxbut i'm having an error on CLI>  the same errors with few other things
02:34.32QwellWhat errors?
02:35.14Dr-Linux1st:
02:35.16Dr-Linux<PROTECTED>
02:35.16Dr-LinuxDec  9 18:29:22 WARNING[23520]: chan_zap.c:10816 setup_zap: Ignoring signalling
02:35.25file[laptop]WARNING is not an ERROR
02:35.50Dr-Linuxyes, but i could be problem later
02:35.51Dr-LinuxDec 10 07:34:29 NOTICE[17962]: rtp.c:1153 ast_rtp_raw_write: RTP Transmission error to 202.125.141.6:8000: Operation not permitted
02:36.23Dr-Linuxactually sir
02:36.44[hC]hmm. yeah. chanspying on cvs head is fine, a little choppy cause im downloading, then when i chanspy on 1.2.1 its so choppy i cant hear anything, and * crashes.
02:36.46NuggetQwell: I doubt you'd want a full res feed.  :)
02:36.47[hC]that sucks.
02:36.56Dr-Linuxi can't see CallderID when someone outside caller calls in via ZAP channels
02:37.03QwellNugget: depends on what you're doing :p
02:37.07Nuggetheh
02:37.37QwellDr-Linux: are you paying for callerid?
02:37.48*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
02:37.57Dr-LinuxQwell: no?
02:38.16harryvvyea, if you dont have caller id service then obviosly you wont see it in cli
02:38.17Qwell...
02:38.26Dr-Linuxi'm using softphone :S
02:38.43Dr-Linuxhhm.. :S
02:38.56*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
02:39.02Dr-Linuxi'm sorry i'm from pakistan, and my server is in US, so i'm not sure
02:39.22Dr-Linuxi thought US all numbers have Caller ID
02:39.27harryvvQwell, this is a issue my wife brought up and yet to find what may be causing it. We have telus with CID and CW but she says since we hooked up the pbx it does not show CIDCW while she is on the phone when someone else is calling in . any ideas what could be causing this?
02:39.46Qwellcidcw is a separate service in some places
02:39.52Dr-Linuxin pakistan, we don't pay for Caller ID, but if someone buy new Phone connection from Telco, there is already Caller ID
02:39.55QwellVerizon charges extra for it...how lame is that?
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02:40.02harryvvqwell, we have cidcw
02:40.39file[laptop]ooh
02:40.41file[laptop]that's hot
02:41.00Dr-Linuxwhen someone calls, i see on the softphone screen "asrecieved"  thats what i defined in the zapata.conf
02:41.01moralecrazy my voip provider is charging me for local calling minutes wtf.
02:41.35Qwellmorale: well...yeah
02:41.44Qwelllocal still costs them.  why wouldn't they charge you?
02:41.59harryvvmorale, how much per min?
02:41.59Dr-Linuxharryvv: so i was thinking, if we have here free Caller ID, then in the US .. VOILA .. but .. amazing
02:42.09morale0.019 cents a minute
02:42.14QwellDr-Linux: callerid costs quite a bit of money here
02:42.17moralei thought it was free local calling.. oh well
02:42.23QwellI think Verizon charges something stupid, like $6/month
02:42.32harryvvthen i wouldnt complain about it morale...its still probebly less then your lec would charge per month.
02:42.47Dr-Linuxhhm.. ic
02:43.06Dr-Linuxharryvv: i thought local call is free in US
02:43.16Qwellnot with voip
02:43.22QwellThere is no such thing as "local"
02:43.38Dr-Linuxhhm.. ic
02:43.41harryvvsince when is LEC free?
02:43.51harryvvlike qwest pots? its not free
02:43.53*** join/#asterisk alephcom (n=alephcom@207.34.97.130)
02:43.54Dr-Linuxbut in my case i don't have any VOIP provider?
02:45.20harryvvI wonder how far a voip termination provider can be before latency becomes a issue.
02:45.34Dr-Linuxi have 2 TDM FXO cards, and 8 lines connected, but i'm not sure it they are free local calls, which dial number is local or what :S
02:45.35Qwellharryvv: I terminate to Oz without much lag
02:45.46harryvvI figure there are a number of variables such at the mount of hops, the bandwith on those links ect.
02:46.04harryvvOz?
02:46.11QwellAustralia
02:46.25harryvvk
02:46.31Qwellymmv
02:46.33harryvvAnd you are where?
02:46.37Dr-LinuxQwell: we have using for our call center Multitech heaving VOIP gateway, Televanage (purchased), so now i installed asterisk
02:46.38QwellUS
02:46.41harryvvk
02:46.49harryvvThat means any country is fair game.
02:46.54Dr-Linuxreally its voice quality and few things are better than ..
02:46.55Qwellwell...
02:47.12Qwellharryvv: depends on your connection, their connection...the connection in between...
02:47.12harryvvsure
02:47.18harryvvyea again, alot of variables.
02:47.19harryvv:)
02:47.29*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM]
02:47.29*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM]
02:47.29*** join/#asterisk juice (n=juice@mo-67-77-176-14.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
02:47.40harryvvI did make contact with a user voip to voip in ireland. it was a pretty good connection.
02:47.44QwellOz for us is easy.  We go straight downward through the earth, and end up there. :D
02:48.14Dr-Linuxharryvv: can you tell me, if Asterisk is free and i found it good, then why peoples are using other expensive things??
02:48.16harryvvI just wished I knew about asterisk 5 years ago
02:48.24QwellDr-Linux: because people are stupid
02:48.35harryvvDr, its complext to configure and almost no one knows about it.
02:48.45*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:48.47harryvvits not a consumer item
02:49.08Nuggetit's really easy to make the wrong choices with asterisk and end up with a crappy mess, too.
02:49.14Dr-LinuxQwell: i'm in the city here almost 1000 call center, but only 1 call center have Asterisk running :S
02:49.19harryvvyup
02:49.46harryvvDr, what do those other 999 call centers use and also..where are you?
02:50.00Dr-Linuxharryvv: pakistan
02:50.07newlQwell: I took the long way here via airplane. :)
02:50.20Qwellnewl: well, you aren't a packet, are you? :p
02:50.24Dr-Linuxthey all give support to US peoples/companies
02:50.38newlI might be rabbit, I might.
02:50.50harryvvDr-Linux where to most of those call centers originate from?
02:50.57harryvvokay never mind.
02:51.12Dr-Linuxharryvv: the call center CSR  take 333$ USD per month, and thats too much ..
02:51.13harryvvDr-Linux you need to know what thay are paying per min
02:51.16Dr-Linuxsorry
02:51.19Dr-Linux300$
02:51.21nassyim trying to understand the difference between PSTN and POTS; T1 and ISDN PRI.anyone know where i can find a beginners explanation to these terms. what is confusing to me is that PRI seems to be used in reference to T1's and i thought POTS connected to the PSTN but i am reading the asterisk handbook and it seems digium makes a PSTN as well as a POTS card
02:51.33Qwell$300/month for what?
02:51.33harryvvDr, what does that equate to per min?
02:51.59Dr-LinuxQwell: get paid, to support US customers ..
02:52.11NuggetI spend more than that a month on beer.
02:52.12QwellThey get that much there?
02:53.11harryvvis that 300 dollars US?
02:53.36harryvvDo you provide customer service ?
02:53.47Dr-LinuxQwell: yes, and thats too much ..
02:53.56Qwelltoo much for what?
02:54.01Dr-Linuxharryvv: yes 300 USD
02:54.11Qwellsomebody dislikes his economy...
02:54.17harryvvSounds like living is cheap in pakistan
02:54.22Nuggetyeah
02:54.24*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
02:54.24Dr-LinuxQwell: too much get paid for one month
02:54.31Qwellwhy is it too much?
02:54.31Dr-Linuxsalary
02:54.36tengulrehi,all
02:54.41tengulreI m coming!! :)
02:54.59harryvvWe have some call centers here in Bc
02:55.01harryvvCanada
02:55.04Dr-LinuxQwell: why too much?  huh .. 300$ USD is very very good pay in pakistan
02:55.12QwellDr-Linux: So, go be a CSR
02:55.18Qwellbe rich...or something
02:55.56Dr-LinuxQwell: i don't wanna be, my field is different
02:56.02Qwellbut you're poor?
02:56.09Qwellsounds like an easy choice
02:56.21Dr-LinuxQwell: every callcenter wants here a Asterisk Solution, but noone knows here
02:56.41Qwellexplain the logic there?
02:57.14Nuggethow can they want an asterisk solution if they don't know about asterisk?
02:57.24Dr-LinuxQwell: well, look, 1$ = 50 RS
02:57.33moraleharryvv: kamloops is the call-center capital.
02:57.35NuggetI think you're just an OSShole who hates to see people buy solutions when they could be using free software
02:57.46QwellI should like...
02:57.51Qwellconvince my boss to let me work from home...
02:57.53Qwellthen move to pk
02:58.00Nuggetheh
02:58.01QwellI'd be like...ubar-rich
02:58.03Qwelluber*
02:58.19Qwellbastard
02:58.20Qwell:P
02:58.23Dr-LinuxNugget: well, they know the name and they know there is an opensrouce, but .. who will configure?
02:58.34Dr-Linuxwho knows Linux
02:58.40NuggetDr-Linux: congratulations, you've just answered your own question.
02:59.36Nuggetplus you've got that whole "LA kills your soul" deal going.  :)
03:00.37QwellNugget: yeah...you've no idea
03:01.27tengulrehi, I want to building a voip platform , I need what hardware?
03:03.10QwellDr-Linux: How much does a mansion cost per month?
03:03.18Qwellwith like...10 bedrooms
03:08.10[TK]D-Fendertengulre : VoIP = Software.  What do you want it to do?
03:09.20Dr-LinuxQwell: sir, sorry can you explain "mansion" word?
03:09.29QwellDr-Linux: huge house
03:09.43Dr-LinuxQwell: huge house per month Rent?
03:09.52Qwellyes
03:09.56Qwellor to own
03:10.16Dr-Linuxhhm... depends on places,
03:10.24Dr-Linuxbut let me tell you .. everage
03:11.25Dr-LinuxQwell: i have very good appartment having 2 bedrooms, and i'm pay 38$ per month
03:11.38Nuggetquite cheap.
03:11.51Dr-Linuxand its best and expensive city in my Country
03:11.59Qwellumm
03:12.02Qwellwhat about a massive house?
03:12.08Qwellwith like...
03:12.09Dr-Linuxour X presedent lives near my apartment
03:12.39QwellThis may take a while :P
03:12.47Qwell26 bedrooms
03:12.47Dr-Linuxlol
03:12.53NuggetI spent $38 on dinner tonight.
03:13.01file[laptop]Nugget: big spender
03:13.02Nuggetand it was cheap and not very good
03:13.02Dr-Linuxooh shit :S
03:13.03QwellNugget: I spend $38 on gas to get home from work today
03:13.26[TK]D-FenderQwell : Ford Explorer? ;)
03:13.33Dr-Linuxoooo :S
03:13.35Qwell[TK]D-Fender: close enough :P
03:13.52Nuggetit was cold this week so I didn't drive anywhere.
03:13.57*** join/#asterisk Jestre (n=ack@dargo.trilug.org)
03:14.06*** join/#asterisk Gerriall (n=NonYa@209.42.198.18)
03:14.06NuggetI went to dinner on monday night and then out again today to get more beer and chinese takeout
03:14.10Dr-Linuxso now it make sense, thats why to many callcenters in this City :S
03:14.17JestreHas anything changed recently about the way the cvs checkouts work?
03:14.25NuggetJestre: yes.  they no longer use cvs.
03:14.39JestreOy
03:14.49Nuggetthey moved to svn
03:14.55QwellThe code is no longer open source.  Sorry
03:14.58Nuggetheh
03:15.10JestreStrange, the zaptel and libpri checkouts updated fine... just the asterisk one failing
03:15.22JestreQwell: Good... save me having to compile it then
03:15.25QwellJestre: it's mirrored every night...it's kind of a hack
03:15.31Qwellshould use subversion now
03:15.51Dr-LinuxQwell: sir currently i have 2 servers, 1 in pakistan and otherone is in California office, we have 4 offices, and i have to configure asterisk for all
03:16.41JestreQwell: Okay
03:16.50JestreLemme see if I can find the new docs
03:18.16JestreSo if I cvs up'd my sources for libpri and zaptel, and built and installed already, should I check out the svn versions and rebuild, etc?
03:18.26Qwellwow
03:18.30QwellNugget: spy is free?
03:18.40Nuggetyes
03:19.33robl^spy?
03:20.45Nuggethttp://natpool.slacker.com:2000/live.html?preset=dialup
03:20.54*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
03:22.57file[laptop]is it reallllllly live
03:23.08Nuggetpretty much, yeah
03:23.18NuggetIM me and see when the window appears.  :)
03:23.21robl^its live!!  and it ate my puppy! blast!
03:24.02Qwell2560x1600?  good lord
03:24.53*** join/#asterisk Assid (i=assid@59.183.1.154)
03:24.54Assidheya
03:25.05file[laptop]oh I know what's happening at Digium...
03:25.32Qwellbackhoe?
03:25.48Assidanyone know if theres a way to get 726/gsm on polycom 501?
03:25.54Assidhey Qwell
03:26.04file[laptop]Assid: no.
03:26.07MstlyHrmlsAssid: no, there isn't
03:26.08file[laptop]Qwell: they're splitting up a box
03:26.23file[laptop]with a knife!
03:26.25file[laptop]muahahahaha
03:26.30MstlyHrmlsAssid: it's G.711 & G.729 only
03:26.37QwellNugget: Can you even fit your arms around it? :P
03:26.52Nuggetit cost the equivalent of 27 months of rent for Dr-Linux.
03:27.05Nuggeter, no, 68 months
03:27.11Qwellyeah...heh
03:27.17Assidi thougth maybe a firmware upgrade or something..
03:27.30Dr-Linuxi got free g729
03:27.35MstlyHrmlsAssid: nope
03:27.47QwellNugget: 30"
03:27.48Qwell?
03:27.52Nuggetyeah
03:28.01Qwellultra-widescreen?
03:28.13Nuggetno, just normal widescreen
03:28.21Nugget16:10 or whatever apple uses
03:28.35Qwell16:9?
03:28.45Nuggetno, they make it a little bit taller to fit the dock  :)
03:28.51Qwelloh
03:29.01Nuggethttp://slacker.com/cinema.php
03:29.06Qwellyeah, saw that before
03:29.28Assidhrmm.. i wonder which takes more cpu power .. 726/gsm
03:30.58*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
03:30.58QwellNugget: sent that link to anybody at apple yet?
03:30.59Assidhrmm.. i cant use SVN on that box anymore.. if i load up subversion, i can never revcompile apache
03:31.01Nuggetheh, no
03:31.07Qwellbet something like that would look great in their advertising, heh
03:31.15Nuggetbut it gets pasted to the apple fanboy message boards all the time
03:32.28*** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
03:32.58AyanoHey all, has anyone in here used or heard of gafachi?
03:33.28SkramXAyano: I have heard of them a bit.
03:33.30SkramXNever used them
03:33.42SkramXWhat seems to be the problem-o?
03:37.18*** join/#asterisk dogtanian (i=dogtania@eye.ham.zee.walroos.com)
03:39.42asterboyanyone selling an X100M ?
03:44.18asterboywhat is the cheapest way to get FXO?
03:44.44asterboyAlso, what is the cheapest termination service?
03:44.51Ayanoasterboy, what are the reasons for fxo?
03:45.11AyanoAnd termination, there are a bunch of cheap ones, but sometimes you get what you pay for
03:45.18asterboyso far because I have some clients I can't get off the pots tit.
03:46.12[TK]D-FenderCheapest FXo would be an X100p clone.
03:46.22[TK]D-FenderWould I suggest it?  probably not
03:46.26asterboyya thats what I figure.
03:46.36asterboyI have one and its working great
03:46.44Assidacutally. i use it. its pretty alrite
03:46.59Assidmy caller id doesnt work tho
03:47.12asterboyinteresting.
03:48.11asterboyAyano, who is offering the lowest price on termination?
03:48.29Assidthe cheapest ive seen is voipjet
03:48.41Jestremake mpg123 still required for moh?
03:48.44Assid1.5 cents
03:49.48SkramXJestre: Yes
03:49.58SkramXAyano: I thought they were 1.3c.
03:50.25AyanoYou can get it for less than a cent i think, but I have seen it a bunch of places for 1.3
03:50.28Assidim not ayano
03:50.34Ayanolol
03:50.44SkramXOh, lol, Sorry.
03:51.22*** join/#asterisk bmg505 (n=leon@dsl-146-63-152.telkomadsl.co.za)
03:51.37Assidthey write 1.3 .. but they bill 1.5
03:52.01Ayanolol,  what company was that Assid?  Is it because of tax?
03:52.41Assidyeah
03:52.44Assidvoipjet
03:52.52*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
03:53.36SkramXHiya, brookshire
03:53.53AyanoAsterboy, how many lines do they need?
03:54.32asterboyI'm reselling, so I would like 10 per month
03:55.41*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
03:55.41AyanoI see.  Then you need incoming as well?
03:55.46Assid10 per month?
03:55.52asterboyyes
03:56.03asterboyDID I can get for $3/month
03:56.10AyanoWhere?
03:56.20asterboyhttp://www.didx.org/did/
03:56.21AssidDID's are officially pretty cheap
03:56.24Ayanoand do they charge incoming minutes?
03:56.25asterboyyes
03:56.35Assidthey charge?!?!
03:56.43Assidi like free incoming
03:56.45Ayanosome do
03:56.48Assidmakes much more sense
03:58.01Assidamazing how UK has cheaper DID's
03:58.03AyanoDo they offer e911 is the next question you need to ask.
03:58.15Asside911  is for outgoing
03:58.45Assidand actually, you could have a way around them, by setting 911 to your local police station
03:59.03asterboynot sure on the charges.
03:59.04Assidor.. if you dont mind investing, buy an FXO and have a landline to it.
03:59.06AyanoAnd no you cant.
04:00.15AyanoYes e911 is for outbound.  If your reselling voip, your lines have to be capable of dialing 911, and it bein traced to that lines address without saying a word.  At least thats the way it is in the states
04:00.39Assidwell.. set the caller id..
04:01.03AyanoYes, but that caller id has to be registered to the correct address.
04:01.07Assidand eithe use a zap channel to make the outgoing call. OR get a provider who lets you make it
04:01.21*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
04:02.00AyanoUsing a zap channel doesn't work because it will come up with the providers address.
04:02.41Assidokay.. so get your provider to support it
04:03.16AyanoYes.  Both your voip, and your zap
04:04.52Assidthere are always ways around things, you could have your end client setup his local police station in the 911 routing
04:05.16Assidso whenebver he dials 911, it dials the subscribers local police station
04:06.22Ayanoyes, it gets to the right place, but do they know where the call came from?
04:06.27brookshirehey!
04:06.27asterboyI got a did for the US and the UK at $1 per month, no per minute charges
04:06.43Assid1 buck?
04:06.50Assidnot bad.. not bad at all
04:06.50asterboyyes
04:06.54AyanoIf the answer is yes.  than your in business.
04:07.18Assidas i said. you can always work something out
04:07.22AyanoAnd one more monkey wrench is that all of your lines have to be able to call 911 all at the same time.  If that makes any sence
04:07.30*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM]
04:07.30*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM]
04:07.30*** join/#asterisk juice (n=juice@mo-67-77-176-14.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
04:07.34Assidyeah
04:07.38*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:07.39Assidjust get good amt of outgoing providers
04:07.47Ayanolol yep
04:07.50asterboyin case katrina comes knocking.
04:07.56Ayanolmao
04:08.08asterboy:)
04:08.12Ayanowhere the hell did you get dids for a buck with no useage?
04:08.20Assiddidx?
04:08.21asterboydidx.org
04:08.25Assidhrmm
04:10.31Ayanovery nice.  I'm going to have to check that out.
04:10.57AyanoI gotta run guys.  It was nice chatting
04:11.29Nuggetmoo  :)
04:11.37asterboynight
04:11.53asterboyday (in case your on the other side)
04:12.18Ayanonight guys
04:12.56Assiddamn.. i dont see a US did for $1
04:13.26asterboylol, not likely...I think thats their test line.
04:13.38asterboyUS is the more expensive for some reason...$4.5
04:13.41asterboyon average.
04:14.00asterboyCanada seems to be about $3
04:14.21asterboyMost of UK is $3
04:14.26asterboySome $1
04:14.46Assidhrmm
04:14.52Assidthey have a toll free number support too
04:14.54asterboySouth Africa is $66!
04:14.58sbingnerlol
04:15.03Assidi wonder if they charge for incoming on the toll free
04:18.27asterboynot sure, I'll ask them
04:18.48Assidimagine
04:19.02Assidfree incoming.. and its freefor the other person to call you
04:20.49*** join/#asterisk Assid (n=assid@203.115.64.62)
04:23.09dogtaniani think you can get free dids for the UK now
04:23.59dogtaniannon-geo ones that is
04:24.02asterboyboy, that guy on asterlink sure does not look happy about his phone bill.
04:24.20asterboysweet
04:24.20Qwellasterboy: eh?
04:24.31asterboyhttp://www.asterlink.com/
04:24.56Qwellthe model?
04:24.59Qwellor whatever
04:25.03asterboyyes
04:25.04asterboy:P
04:25.21asterboylooks real serious anyway.
04:25.43asterboyI'd use their service, but I want a girl with big tits on there.
04:26.17asterboywhats happened to our society? all going down hill now.
04:26.22asterboy:P
04:28.05Qwellasterboy: oh, that's there
04:28.10QwellIt's after you login
04:28.16asterboylol
04:28.33*** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
04:28.49asterboyso asterlink is chargin .02/min
04:29.19harryvvasterboy who?
04:29.31harryvvhe is charged .02 per min?
04:29.52asterboyhttp://www.asterlink.com
04:30.07asterboyThat is what they charge for VOIP termination.
04:30.36asterboynot sure if its in, out or both.
04:31.04asterboyI'd like to get VOIP termination with unlimited talk.
04:31.13asterboyjust a fix monthly rate
04:33.03asterboyhttp://les.net/ has 800#s for $2.50/mth but .04/min charge
04:33.15*** join/#asterisk jahani2 (n=k@adsl-128-49-192-81.adsl.iam.net.ma)
04:34.36*** join/#asterisk docelm0 (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net)
04:34.44docelm0YIPPIE!
04:35.13*** join/#asterisk dokhench (n=dochench@adsl-156-13-196.bna.bellsouth.net)
04:35.16harryvvhi docelm0
04:35.33docelm0Sup dude
04:35.48harryvvhe is complaing on that rate which im also charged. bit deal!
04:35.55harryvvohh nota much
04:37.02docelm0I FINALLY GOT MY GD SPA3000 FXO port to work!
04:37.08Qwellasterboy: it's in and out
04:37.09docelm0Im so happy..
04:37.25harryvvyea
04:37.26harryvv:)
04:37.39harryvvpeople dont get to excited about telephone talk
04:37.44docelm0and I found out I will have my dCAP cert in hand soon..
04:37.47docelm0YAY!
04:37.51Qwellharryvv: You're talking to the wrong people then. :P
04:37.52docelm0hehe..
04:38.03harryvvqwell, thay are curios but thats about it.
04:38.13docelm0I got a guy in my office hooked on asterisk.. He likes it so much he went to Miami to the bootcamp
04:38.14blitzragey0 y0!
04:38.17Qwellblitzrage: !
04:38.34harryvvhard to get excited over that. now, take a Toyota Land cruiser diesel im looking at buy it cheap rehab it then its a talked about vehicle
04:38.36moraleasterisk bootcamp?
04:38.38fileblitzrage: !!!
04:38.50Qwellfile: stop outdoing me
04:38.56filetoo bad
04:39.06docelm0Sokol and Assoc puts on a Asterisk Training Bootcamp..
04:39.09docelm03000 buks
04:39.25fileI expect... MADNESS!
04:39.28bsdfreakheh
04:39.35harryvvfor how many hours/days
04:39.50*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:40.16asterboydam...just found yet a cheaper phone company
04:40.21fileanyway, brb from laptop
04:40.22asterboyhttp://acanac.ca
04:40.23blitzrageits 5 days long (bootcamp)
04:40.38blitzragedays are usually about 10 hours long
04:40.43asterboythat will beat any of them so far...$20/month
04:40.54Qwellasterboy: you want a crappy provider?
04:41.02docelm0asterboy who?
04:41.02asterboyare they bad?
04:41.07asterboyhttp://acanac.ca/Features.htm
04:41.10Qwellprobably
04:41.19QwellCheap, good, has support
04:41.21Qwellpick up to 2
04:41.33docelm0I have all 3
04:41.36*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
04:41.36docelm0Plainvoip.com
04:41.48blitzragemy iPods HD is dieing
04:42.00file[laptop]blitzrage: what?!?
04:42.21blitzrageyah -- apple hardware sucks ass
04:42.27blitzrageit makes a wierd noise
04:42.33blitzrageand I got an error the other day
04:42.37blitzragegotta try and get it replaced....
04:43.04MikeJ__yes.. it's all blitzrage's fault
04:43.08MikeJ__I agree
04:43.16blitzragetotally
04:43.22QwellI'm balding
04:43.25Qwelldamnit blitzrage
04:43.30blitzragemy fualt?
04:43.46asterboydocelm0, interesting most voip terminators are charging /min....I want a fixed rate per month.
04:43.55distortionanyone have advice on how to setup sip/extensions.conf to allow all users from anotehr sip proxy defined by ip?
04:44.14MikeJ__asterboy, sure... especially if it is unlimited concurrent calls...
04:44.19asterboyacanac.ca is offering $20/month unlimited talk for a FULL service line.
04:44.38distortionright now the proxy sends each individual user in the invite causing asterisk to reject
04:44.54asterboyMikeJ__ how do you mean "concurrent calls"
04:45.01Qwellasterboy: multiple calls at once
04:45.05MikeJ__asterboy, more than one call at a time
04:45.12MikeJ__as in concurrently
04:45.27docelm0asterboy um, I dont know anyone flatrate unless you purchase their ATA and such.. like Vonage
04:45.41asterboythats what I thought you meant...but that would be for example 2 phone lines, no?
04:45.57docelm02 phone lines for what?
04:46.06docelm0or what do you mean by 2 phoen lines?
04:46.10asterboyconcurrent
04:46.13docelm0Nope..
04:46.24MikeJ__well.. I think broadvoice or voicepulse does.. but they do it based on unlimmited for the first concurrent call.. if you do more that 1 call at a time, it cost above and beyond
04:46.24asterboylike a conference call
04:46.29MikeJ__I don't recall which one
04:46.40docelm0based on minute.. Unlimited calling cause you pay by minute.. For a monthly yes.. normally its limited to 2 concurrent calls at a time
04:46.43file[laptop]broadvoice... does it
04:46.45MikeJ__asterboy, well.. a sip account is not like a pots line...
04:46.51file[laptop]but, who knows how well it'll work for you
04:46.55MikeJ__you can make more than one call at a time
04:47.10asterboyfor conferrencing?
04:47.18MikeJ__for calling!
04:47.52dogtanianif i want to recieve more than one call at the same time (ie. have have a few POTS callers in a queue) do I need more than one DID?
04:47.54MikeJ__at an office, bob, jane, and dick are all making outbound calls at the same time
04:48.07MikeJ__dogtanian, no
04:48.08asterboyah
04:48.11QwellMikeJ__: I took away Bob's outgoing context rights...
04:48.22MikeJ__Qwell, BASTARD!
04:48.22asterboyI was thinking 1 person making multiple calls.
04:48.23MikeJ__:P
04:48.24dogtanianMikeJ__: ah good - that's what i wanted to hear
04:48.26asterboygot you now.
04:48.31Qwellasterboy: that too
04:48.38asterboyah
04:48.44asterboyok makes sense.
04:49.02MikeJ__dogtanian, that being said, you need to have some way to receive multiple calls... a sip account that lets multiple inbound, or a pri, or somthing
04:49.11file[laptop]mmm PRI
04:49.19MikeJ__yeah.. I'm all about PRI's
04:49.26distortiondid someone just grab my PRI?
04:49.31MikeJ__yes
04:49.31file[laptop]MikeJ__: excuse me but your PRI is showing
04:49.31asterboywell $20 unlimited per month for anywhere is dam good.
04:49.39distortionyou grabbed my PRI!
04:49.42MikeJ__yes
04:50.03distortion./M MikeJ__ that felt good
04:50.06distortionerrr...
04:50.10MikeJ__uhhhh
04:50.14MikeJ__bad touch
04:50.38file[laptop]you two should get a p2p link!
04:50.42file[laptop]much more private
04:50.49Qwellmmm
04:51.11QwellI wonder how much Verizon would charge me to get a PRI from my bedroom to my living room
04:51.37MikeJ__Qwell, why would you need verizon to run a piece of cat 5 in your house/
04:51.38MikeJ__?
04:51.41distortionQwell, ill install one for tree fiddy
04:51.46Qwellno, must be T1
04:51.52QwellPRI
04:51.55dogtanianMikeJ__: i was thinking of building an * box for my house, and using this service http://www.voiptalk.org/products/VoIPtalk+Services/iaxtalk
04:51.56MikeJ__ummm....
04:52.08MikeJ__dogtanian, congrats!
04:52.12distortionpri runs over cat5 :)
04:52.38dogtanianMikej__: it looks as if it'll allow me to take multiple incomings from POTs, although it doesn't say
04:52.49MikeJ__ummmm
04:53.07MikeJ__multiple incomming calls from POT's?  huh
04:53.19MikeJ__your talking nonsense ...
04:53.19dogtanianyeah. simual calls
04:53.26MikeJ__on a pots line...
04:53.28MikeJ__cool...
04:53.34dogtanianheh. forgive me. i'ma  noob to all this
04:53.34MikeJ__get me one of those.
04:53.40file[laptop]just nod your head
04:53.58file[laptop]exactly.
04:53.58MikeJ__wassup file..
04:54.07file[laptop]watching Mr. Holland's Opus
04:54.23MikeJ__and typing at the same time
04:54.26file[laptop]and wondering why a support agent sent me trouble tickets, that I don't handle...
04:54.33harryvvwhats the going min/rate for sat phones?
04:54.34MikeJ__bastards
04:54.34file[laptop]when one is from 20 days ago...
04:54.41file[laptop]and marked urgent
04:54.42file[laptop]LOL
04:54.53MikeJ__heh
04:54.54MikeJ__oops
04:55.24asterboydogtanian, that iaxtalk seems more geared for the UK...anything like that in Canada?
04:55.39file[laptop]MikeJ__: I'm tired, tell me a story Daddy Jerris!
04:55.40MikeJ__I've spent the day beating up on segfaults and crazieness..
04:55.46MikeJ__awwww...
04:56.11MikeJ__once upon a time I was trying to talk on the phone w/ tony while he was in a very very loud starbucks...
04:56.16MikeJ__and it didn't work...
04:56.24file[laptop]the end?
04:56.27MikeJ__so he went to sit in his car in the cold
04:56.29docelm0woo HOO! I got it.. PSTN setup for incoming and ALL LD is going to the company I work for.. Life is good!
04:56.34MikeJ__and coded from his car
04:56.36MikeJ__the end
04:56.55distortionbut what did he code?
04:57.00MikeJ__ummmm
04:57.02file[laptop]madness
04:57.02distortionwe want more!
04:57.09MikeJ__tonight?
04:57.09dogtanianasterboy: not sure tbh. i'm actually in london so it suits me :)
04:57.09Qwellapp_starbucks
04:57.29MikeJ__naw....
04:57.38MikeJ__super duper secret stuff
04:57.42docelm0hehe
04:57.56MikeJ__he tried to make what I wrote stop making bad bad segfaults!
04:57.59MikeJ__ouch
04:58.13file[laptop]bad MikeJ__ with segfaults and stuff
04:58.15file[laptop]ALTHOUGH
04:58.22MikeJ__hmmm
04:58.24file[laptop]I've managed to make codec_ulaw segfault today
04:58.31Qwellfile[laptop]: you rock
04:58.33MikeJ__file, have tou even look at that stuff yet?
04:58.33file[laptop]which is funny
04:58.40MikeJ__file, with coremedia?
04:58.56file[laptop]MikeJ__: looked at it? I wrote it!
04:58.59dogtanianasterboy: my objective is to have multiple people dialing-into a single non-geo number, and being kept in a queue if the ip-phones are busy
04:59.10file[laptop]but yeah I converted a few test codecs over to it
04:59.11MikeJ__yeah.. been watching tthe commits
04:59.19file[laptop]they work fine
04:59.25MikeJ__no, I mean that other stuff
04:59.30MikeJ__the super secret stuff
04:59.39file[laptop]the one I added into the Makefile?
04:59.50asterboydogtanian...are you building for business, personal or resale?
04:59.50file[laptop]but didn't actually "commit"
04:59.58MikeJ__umm
05:00.00MikeJ__heh..
05:00.07MikeJ__yeah.. I saw that.. that was funny
05:00.12MikeJ__ummm..
05:00.15MikeJ__no, not that...
05:00.16dogtanianasterboy: personal to beging with, but i may use it for an internet shop i'm putting together
05:00.24MikeJ__but kinda related...
05:00.29MikeJ__the one I wrote yesterday
05:00.46asterboyya...thats where I see a lot of this telephony going...telcos being split up by little guys like you and me.
05:00.56file[laptop]I haven't been following what you've been up to, well, I've been looking on Tony's terminal every so often
05:00.57dogtanianasterboy: i juts thought i'd sit on this chan for a bit to find out whether * will be worth the effort for me
05:01.12asterboy* rocks
05:01.13MikeJ__file, yeah.. that stuff
05:01.23dogtanian:)
05:01.24asterboyespecially on linux
05:01.34dogtanianyeah
05:01.37asterboycause linux rocks
05:01.47file[laptop]MikeJ__: ahhhh you've been pretty quiet lately... up to no good I see indeed yes...
05:02.03MikeJ__yes.. yes indeed
05:02.06dogtaniani'll prolly try *@home first coz it seems pretty simple to set up... and maybe buy a couple of cisco handsets on ebay
05:02.16MikeJ__pull it down off that your internal svn
05:02.36MikeJ__pull that down off your internal svn that is
05:02.41asterboygo for the polycom stuff...better prices and no yearly fee
05:02.56dogtaniando the cisco ones have a yearly fee?
05:03.08Qwellonly if you want to have legal right to use the firmware updates
05:03.10asterboynot sure if all do
05:03.12file[laptop]MikeJ__: I just may
05:03.28dogtaniani saw somethign about licensing, but i didn't bother to read the detail
05:03.31dogtanianhmm
05:03.39asterboyignorance is bliss
05:03.42QwellI'm fairly sure the license is only for ccm
05:03.46dogtanianyeah
05:03.51Qwellbut, you do need a valid smartnet contract
05:03.56Qwellin order to get firmware updates
05:04.13QwellIt's like $10-15 or so.
05:04.18dogtanianah that's ok
05:04.19moraleis there a way to make it so if my internet connection goes offline it rings my SIP/Zap lines in my house to let me know? :)
05:04.24Qwellwith polycom, don't you have to like...
05:04.27moralei need to write app_cronjob.c for asterisk
05:04.33Qwellconvince somebody to give you the firmware?
05:04.50dogtanianit would be pretty sucky for them to ask for any more than than considering that hardware isn't exactly cheap
05:05.21dogtanianit's be like me having to pay a monthly fee to watch my television... uhmm.. hang on... :)
05:05.40Qwelldogtanian: welcome to the UK
05:05.43dogtaniani've been warned away from the Grandstream's
05:05.46Qwellyeah
05:05.49dogtanianQwell: yeah exactly
05:05.55Qwellor, the US
05:06.03Qwellwhere in order to get a reasonable amount of channels, you need cable
05:06.11dogtanianIn the US it isn't a TV license per se tho is iy
05:06.17dogtanian*it
05:06.24Qwellnot quite
05:06.25dogtanianmore like subs
05:06.48dogtanianand they don't throw you in the brink if you don't pay.. i guess you just get cut-off
05:06.56Qwelltrue
05:07.44dogtanianwell, i reckon the cost of a decent cisco is worth the investment
05:07.57Qwellit is, imo
05:08.13dogtanianand the one with the colour touchscreen looks well sweet
05:08.18Nivexmorale: why not just use the system's cron and drop a call file in /var/spool/asterisk/outgoing ?
05:08.21dogtanianalthough i'd prolly drool all over it and break it
05:08.25Qwellyeah...no sip firmware for the 7970
05:08.27file[laptop]I like my 7960G...
05:08.29Qwellbut, it rocks with sccp
05:08.38Qwell<3 sccp <3
05:08.43moraleNivex: that would work too
05:08.47dogtanianah i didn't realise. no sip :/
05:08.51QwellI converted my personal 7960 to sccp the other night, heh
05:09.51dogtanian:)
05:09.57dogtanianwhich gateway do you use?
05:10.01Qwellgateway?
05:10.05SkramXWee.. compiling kernel for gentoo
05:10.22harryvvsixtel is ticking me off. party on other end says my call is crappy and can only hear a cylibal here or there.
05:10.28SkramXQwell: Why would you want to use SCCP
05:10.28dogtanianuhm.. i don't know the vocab yet :) .... which provider do you use to dial out to pstn?
05:10.33QwellSkramX: because sccp rocks
05:10.37SkramXharryvv: yes, Sixtel basically sucks.
05:10.40Qwelldogtanian: asterlink and nufone
05:10.46MikeJ__harryvv, then don't useem
05:10.46harryvvis there a way to force a call to another voip carrier using the phone?
05:11.11harryvvsixtel quality has really improved
05:11.15harryvvbut not tonight.
05:11.29MikeJ__harryvv, huh?
05:11.36SkramXLOL
05:11.37MikeJ__force using the phone?
05:11.41file[laptop]it's the night... of DOOM
05:11.43MikeJ__you mean during the call
05:11.46SkramXWhatever, bro
05:13.14JestreWhat is the System Menu stuff on Asterlink Extreme?
05:13.24QwellJestre: ask file
05:13.27Qwell:P
05:13.28harryvvno not during the call thats pretty much impossible but say make a second call and using dtmf sequence before the call would switch to another service.
05:13.34file[laptop]yes, ask me... when it's not 1AM please
05:13.34SkramXJestre: #asterlink
05:16.51asterboywow unlimited voip for $15/mth
05:17.07MikeJ__unlimited...
05:17.25asterboyno per min charges, calls in/out anywhere in NA
05:17.54asterboyDIDs with per min charges are about the same with just 500 min.
05:18.15ManxPowerAnyone that believes unlimited means unlimited is a fool
05:18.29file[laptop]very true
05:18.48asterboyVonage has unlimited for $40
05:19.11asterboyI left the phone off the hook and came up with almost 3000 min...$40
05:19.32*** join/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net)
05:19.42h3x0r4t0rAstinus: so it cost the same as voipjet
05:19.47h3x0r4t0r$0.013/min
05:19.57qw3rtyanyone here using a cisco phone via skinny?
05:20.00harryvvasterboy wow
05:20.04MikeJ__nighty night...
05:20.09asterboynight Mike
05:20.26asterboynow if I can do that with a $15 account...wow
05:21.43asterboyhttp://www.ordervoip.com/compare.php?R=127788
05:22.24harryvvsun ulta fire 1000 had 5x the power of todays comparable servers and is 1/5 the power required to power it.
05:22.32harryvvhad is has
05:22.33harryvv:)
05:23.47Qwelland only 300x the cost
05:23.50ManxPowerUm, 3000 mins is not all THAT much,.
05:24.04QwellManxPower: you should see my nufone account
05:25.32asterboythat was for the one call
05:25.51QwellManxPower: mind a quick msg?  I want to show you this
05:26.01Qwell3 lines...no reply needed
05:26.05ManxPowerQwell, does it require thinking?
05:26.08Qwellnone
05:26.08moralesun hardware is way overpriced.
05:26.13ManxPowerQwell, OK.
05:28.43*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
05:29.33sylemorale alot of us realize that 5 years+ ago :)
05:29.48QwellI don't know...
05:29.49pauldyhrm anyone know how to work with predial on 1.2
05:29.56Qwellthat new thing Sun is doing with the support contracts is kinda cool
05:29.59*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
05:30.11pauldyI have it working fine on 1.0.9 but for some reason 1.2 is not liking it much
05:30.17Qwellfree opteron system, if you get a $30/mo support contract for 3 years
05:30.33qw3rtyWhen I define a phone in the skinny.conf I put a context line in there.  Does the corresponding context needs to go in the extentions.conf?
05:30.47Qwellqw3rty: of course
05:30.52*** part/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
05:30.58SedoroxQwell: hell.. I'll do that.. my $80/month payments for training I never used is coming to an end :p
05:31.04Qwellheh
05:31.07sylesimple fact is you can get 3-4 linux system for the price of a Sun
05:31.12QwellSedorox: it actually is a decent system
05:31.15Qwellsyle: not really
05:31.19syleits not efficient
05:31.20Sedoroxany opteron is...
05:31.22Qwellsyle: $800 for an opteron (without the support contract)
05:31.26Sedoroxsun's are nice....
05:31.33QwellSedorox: there was a link on /. the other day, if you're interested
05:31.41Sedorox*looks up at his sun fire x4100 centerfold*
05:31.45Sedoroxhmm
05:31.53QwellI love sun hardware
05:32.07Qwellmy router has never crashed*
05:32.13Sedoroxhehe
05:32.17qw3rtyQwell, ok  If my context line reads  "context=cisco12sp"  should the context in extentions.conf read [cisco12sp] or [ext-cisco12sp]?
05:32.24Qwell(*the driver for the happymeal has...on multiple occasions)
05:32.40sylei'll beleive 800 when i see it hehe
05:32.45Qwellsyle: want a link?
05:32.49sylesure
05:33.01Qwellsec
05:33.12QwellI think it was tuesday...lemme find it
05:33.56*** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
05:34.03qw3rtyQwell, did you see my question above?
05:34.19Qwellqw3rty: the same as any other channel driver
05:34.38SedoroxI'm in love with Sun's 19" flat screens
05:34.43qw3rtyQwell, sorry I am still new to this and using asterisk @home right now
05:34.44Sedoroxand even more so with their 21"
05:34.57Qwellhttp://www.sun.com/desktop/workstation/ultra20/
05:35.03QwellSedorox, syle: ^
05:35.13NivexI'm having some trouble checking out gastman from svn
05:35.25pauldyhrm no one know what changed with the way early dial was handled from 1.0.9 to 1.2.
05:35.28Sedoroxyummy
05:35.29sylestarts at a list price of US 895
05:35.39Qwellso I was off by $95
05:35.51QwellThat's for an opteron u20
05:35.55Qwellcome on
05:36.13moralethats bare bones probably
05:36.27QwellThat's for a full system.
05:36.47Qwellhttp://store.sun.com/CMTemplate/CEServlet?process=SunStore&cmdViewProduct_CP&catid=132680  jesus christ
05:36.57QwellI'll admit, THAT is a little expensive :P
05:38.02Nivexsvn: REPORT of '/svn/gastman/!svn/bc/54/trunk': 400 Bad Request (http://svn.digium.com)
05:38.07syleyep rediculous, scares me the most is who knows what they;ll charge for next
05:38.19sylehell at one point you had to buy the cc compiler didn;t you
05:38.52sylei remember SGI's with irix had that , or you could just use gcc but that screwed alot of stuff up
05:39.37moralesgi still makes nice hardware. but irix has not improved.
05:40.42SedoroxI would be interesting to see what the performance different from say a Tyan mobo to a Sun mobo is...
05:40.57sylei think simple fact of the matter is people will compromise abit on stability for pricing
05:41.07syleor we would all be running sparcs
05:41.31Sedoroxlol
05:42.17Sedoroxno legacy
05:42.17Sedoroxnice
05:43.10pauldybah figured out my problem early dial and 9|. with BV does not get along
05:44.25Sedoroxthats not bad tho
05:44.31Sedorox$30/month for 3 years...
05:44.40Qwellnot at all.  only comes out to $200 more
05:44.48Sedorox$1080...
05:44.50Sedoroxyea
05:44.57Qwell1078, technically
05:45.15Qwellon, wait
05:45.19Sedoroxbut its a Opteron 144, ATI (ew) 512 ram, 80gig drive, gig eithernet... firewirell usb.. lots of PCI-Express choices...
05:45.22Qwellit's 3 anual payments
05:45.25Qwellannual
05:45.28Qwelloh
05:45.36Sedorox30*36
05:45.36Sedorox:p
05:45.56Qwell"Payment for this support will be billed annually with the first payment due upon ordering this subscription. The remaining amount billed in two installments due at the beginning of year 2 and year 3."
05:46.02Qwellstill
05:46.26Sedoroxso $360 upfront
05:46.40syleqwell i think this is page you wanted btw
05:46.46sylehttp://store.sun.com/CMTemplate/CEServlet?process=SunStore&cmdViewProduct_CP&catid=131490
05:46.57Qwellsyle: nope
05:47.06Qwellhttp://store.sun.com/CMTemplate/CEServlet?process=SunStore&cmdViewProduct_CP&catid=133232&PROMO
05:47.08QwellThat one
05:47.28trixter~seen benjk
05:47.32jbotbenjk is currently on #asterisk (5h 32m 42s)
05:47.43trixterthought that logged last spoke time
05:49.36*** join/#asterisk Lurr (n=pr0ph3t@pcp04927291pcs.wolfrd01.fl.comcast.net)
05:51.00sylehmmm
05:51.15syleforced into 3 year contract
05:51.27*** part/#asterisk Lurr (n=pr0ph3t@pcp04927291pcs.wolfrd01.fl.comcast.net)
05:51.31syle29.99*36 is over a grand
05:52.00Sedoroxsyle: read up
05:52.00Sedorox:p
05:52.02Sedorox$1080
05:52.16QwellIf you're going to get a support contract from sun anyways...
05:52.30sylewell 1079.64 to be exact
05:52.46Qwell1078 to be exact
05:53.23syleon link you gave?
05:53.32syletry your find button hehe
05:53.32Qwellthink so
05:53.41Qwell1,078
05:54.12syleok am i on drugs can someone else go to last link qwell posted and tell me if it says 1078
05:54.21syleanywhere at all on the page
05:54.29Qwell3 Year Service Support Subscription   [+ $1,078.00]
05:54.42Qwellsays it on the page twice
05:54.46QwellSubtotal   $1,078.00 List Price
05:55.13sylehttp://store.sun.com/CMTemplate/CEServlet?process=SunStore&cmdViewProduct_CP&catid=133232&PROMO
05:56.11syleregardless you obviously smoke some weed tonight and are on different page, don;t matter, but i wonder how popular it will be
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05:56.29Qwellsee the "select" button?
05:57.01syleyes i see its on the select button link not the original link posted
05:57.14QwellWell, $0.00 < $1.078 :P
05:58.01Sedoroxits nice for someone like me who might be able to do $30/month.. but not $890 one time
05:58.18sylei use to to a sys admin for space solaris boxes back in '99 i think
05:58.24sylemost stable boxes i ever worked with
05:58.27Sedoroxhehe
05:58.36SedoroxI dunno what my school has in the sun lab....
05:58.40Sedoroxthey are ok machines..
05:58.40syles/space/sparc
05:59.06sylebut i also noticed only million dollar companies had them
05:59.31sylefiber channel arrays, veritas etc
06:00.00syleveritas was used to backup fs , oracle db etc
06:01.04sylebut anyways i fucked the boss's girl he wanted and lost that job, but was still a good experience on waste of money hehe
06:02.32Sedoroxhttp://pclug.pct.edu/~penbra67/Sun.jpg
06:02.34SedoroxI love that screen
06:02.53Sedoroxthats oe of the stations in our sun lab... hooked into my laptop (the only one who could do it since my laptop has DVI out)
06:03.22sylehmm looks exactly like the samsung syncmaster 191T i;m working on right now
06:03.27Sedoroxhehe
06:03.27SedoroxI like it.. 'tis a nice screen
06:04.20syleyeah its nice, got a burnt out pixel though blah
06:04.48Sedoroxlol
06:04.51Sedoroxits weird...
06:05.01Sedoroxthe Gateway LCD's that they are have in one of th other labs here
06:05.04Sedoroxis actually burnt in
06:05.11Sedoroxits got a burn in of the novell login screen...
06:05.16SedoroxI didn't think that happened with lcd's
06:05.18sylewhen it comes to lcd screens, i will never buy one again unless i buy the floor model and see their are no burnt out pixels
06:05.44sylebastards won;t replace them unless you have at least 6-10 burnt out
06:06.14Sedoroxhehe
06:06.26Sedoroxmy parents bought one
06:06.30sylethere are programs to fight back against that, to purposely help you burn out more to send it back but they are difficult to find
06:06.33Sedoroxthe one side of the screen is covered in them
06:06.49Sedoroxbut you really don't notice unless its a black screen (they run windows.. how often does that hapen :p)
06:06.56Sedoroxor if your really looking for it.. so I didn't push for it
06:07.06Sedoroxinteresting
06:07.17syleto me all the time, i am always in securecrt on my linux boxes lol
06:07.28sylealways black, i;ve come to live with it now though
06:08.19Sedoroxlol
06:08.43sylebut exactly right, i;ll have some nice black background for a website to look at to load on floor model before i buy another one lol
06:09.59sylesedorox
06:10.07syledo you know anything about plasma?
06:10.20syleever hooked up your computer to a big plasma screen?
06:11.08Sedoroxno :-/
06:11.14Sedoroxhaven't delat that much with HD stuff yet...
06:11.18Sedoroxbig plasma or big lcd
06:11.48sylei saw it at one guys house, but it didn;t look that great, wondering if their is a quality difference in plasmas, was difficult to make out anything still on small fonts
06:12.10Sedoroxwell plasma's I think take a bit to warm up (dunno how long he had it on)
06:12.18Sedoroxalso... was it all stock settings? did he tune it?
06:12.29syledoubt it
06:12.49sylebut i am convinced he ran rca instead of digital to it
06:13.45sylecan;t say for sure but that is what i thought after i left
06:14.23sylei think a plasma for a workstation is just to damn big hehe
06:14.48Sedoroxlol
06:14.49JonR800there's not much of a point.. it's not like you're getting more workspace, in fact you're probably getting less.
06:15.02Sedoroxwell if he used it on his computer.. the other thing.. was he running in the native res?
06:15.02sylei don;t think anything over 19 inches is a good idea or it really fucks you up for developping web applications at a resolution everyone else uses
06:15.07JonR800a 1080p display might now be bad..
06:15.10Sedorox21" is nice...
06:15.16Sedoroxthe 30" apple's are really nice :p
06:15.19JonR800now=not
06:15.21sylei don;t think most people use 21's
06:15.35syleso you may not scroll on a webpage where they would
06:16.06sylebut hell why not go 21 inche, 56 inch plasma etc if your gaming :)
06:19.43Sedoroxtrue.... but eh... I would love to have 2x 30" cinema displays..
06:19.52Sedoroxbut even 3x17 or 19 is fine
06:20.00Sedorox<--- multi-monitor lover
06:20.38syleyeah i;m thinking of adding a third monitor would really help
06:21.00SedoroxI'm only on my laptop
06:21.11Sedoroxmy vid card doesn't like doing dual with the LCD and a external
06:21.12Sedoroxon linux
06:21.15Sedoroxwindows is fine.. 'tis weird
06:23.19sylei think windows is the best "client" for a workstation so i use that, and linux best as a "server", so i use them to run inet stuff etc, always better driver support in windows unfortunately
06:23.47sylehence why i sit in securecrt alot :)
06:24.17Sedoroxlol
06:24.29syledon;t get me wrong though i did run freebsd as a client in X for like 2-3 years straight
06:24.32sylereally no fun
06:24.53Sedoroxhehe
06:25.01SedoroxI like linux for my personal workstation os...
06:25.09Sedoroxlinux for server.. freebsd for anything network related
06:25.18Sedoroxand windows for people who I just don't feel like supporting
06:25.18Sedorox:p
06:26.03sylewell my thoughts on that are freebsd for single processor machines, linux for dual
06:26.28sylei beleive fbsd doesn;t make enough efficient use of SMP
06:26.38Nuggetneither do most users, either.  :)
06:26.39Sedoroxhehe
06:27.33NuggetI use freebsd for machines that matter, os x for machines I have to sit at, and linux for machines that need to run tuxracer.
06:27.48Sedoroxlol
06:28.01sylenugget are you running asterisk on fbsd?
06:28.04*** part/#asterisk alephcom (n=alephcom@207.34.97.130)
06:28.05Nuggetyes
06:28.10syle5.x?
06:28.11SedoroxI use to
06:28.12Nuggetyes
06:28.22sylehow is it compared to linux?
06:28.33Nuggetzaptel blows goats in freebsd, but otherwise it's fine
06:28.33syleany noticeable differences?
06:28.39Sedoroxhehe
06:28.41Sedoroxwas just gonna say
06:28.45syleyeah taht is what i was worried about
06:28.46SedoroxI had problems with zaptel locking kernel up
06:28.53Sedoroxwell... when asterisk was stopped...
06:29.05Sedoroxbut other then that.. was great...
06:29.13sylethats not great when your running a channel bank or PRI hehe
06:29.21Sedoroxwas only one release of zaptal I had that problem with tho
06:29.26Nuggetyeah, my box with the tdm400p is a linux box
06:29.30Sedoroxlol
06:29.49Sedoroxwe had zaptel for ztdummy... and the other bozes has X100P's in them
06:30.05sylei haven;t upgraded freebsd at home in a long time think i;m still have fbsd 4.3 on it
06:30.27NuggetI've moved most of the distributed.net boxes to 6.0 now, which is really solid.  Much nicer than 5.x ever was.
06:30.32Nuggetbut my home server is still 4.x
06:31.11sylei gave 5.x a shot when it was out for first month, and went right back to 4.x, half the ports were not even compiling etc
06:31.14Sedoroxmy gateway is still 5.3-release
06:31.15Sedoroxlol
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06:32.08sylethe reason i wanted to move to 5.x off 4.x anyways was cause of the wireless card support
06:32.16syleactually supported 32 bit cards in 5.x
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06:33.18sylebutmy favorite part of fbsd is the ports collection so that was the deciding factor to go back to 4.x till it was better tested lol
06:33.45syleprobably yours to , ports and build worlds
06:34.32Sedoroxhence why I use gentoo
06:34.32Sedoroxemerge
06:34.33Sedorox:p
06:34.42Sedoroxsame idea
06:34.49SedoroxI still gotta get the hang of fbsd...
06:34.56Sedoroxbetter then I was with it :p
06:38.43wshsgot me asterisk configured for broadvoice. worked fine for months. couple weeks ago, inbound calls started getting busy. however, i didn't know until recently. when using hardware to connect, inbound calls work fine. does broadvoice no longer play nice with asterisk?
06:38.54syleit just makes administation easier
06:39.17syleif you ever at a job taking care of tons of linux servers try to convert them to fbsd :)
06:39.34Sedoroxif they run fedora I would switch them to gentoo
06:39.35Sedorox:p
06:40.26sylei hear debian is alot like fbsd it has a good packaging system
06:40.43Sedoroxahah
06:41.00sylei use fedora and redhat myself
06:41.04Sedoroxgentoo is probably the most like fbsd in package management
06:41.06SedoroxBAH!
06:41.09Sedoroxthey both suck
06:41.10Sedorox:p
06:41.14Sedoroxin my opinion anyway
06:41.15syleso use to working at companies over the years that ran redhat
06:41.31syleuse to /etc/sysconfig style scripts now and come to like them
06:41.38Sedoroxlol
06:41.48Qwellit's all about /etc/conf.d/
06:42.01Sedoroxyup yup
06:42.19syleto me its all about rc.local
06:42.25sylei started with slackware 3.0 lol
06:42.31Sedorox<PROTECTED>
06:42.31Sedorox:p
06:42.40SedoroxI started my linux world with Redhat 5.1
06:42.47Sedoroxthen 6.1... then mandrake 6.1
06:42.50Sedoroxthen slack 7
06:43.01Sedoroxmoved on slack till 9... then went and currently with gentoo
06:43.04QwellI like gentoo, because there is no "upgrading"
06:43.11Qwelllike...no distro version
06:43.14Sedoroxhehe
06:43.17Qwelljust the profile, but that's damn easy
06:43.18Sedoroxtrue...
06:43.32QwellIt is a complete bitch to upgrade remote servers from FC1 > FC2, etc
06:43.42sylei like how i can do yum update kernel on fedora and not worry about kernel upgrades
06:43.57Qwellgenkernel
06:43.59Sedoroxlol
06:44.02Qwellif you must
06:44.04sylehandy
06:44.05SedoroxI roll my own kernels
06:44.08Qwellindeed
06:44.13SedoroxI don't trust distro-made kernels
06:44.14Qwelldistro kernels are bloated
06:44.29Qwelloften patched to hell too
06:44.30Sedoroxmy kernels....
06:44.36syleyeah well you;ll stop doing that one day when you run a system with 100 linux machines and want to upgrade the new kernel everyone month with 100 other things you have to do
06:44.44Qwellsyle: gentoo
06:44.55Qwellmakes that damn easy
06:45.00sylevery nice
06:45.04QwellI think you can make a kernel package
06:45.41Qwelljust have one cluster/machine do all your compiling/packaging, and have all the others point to it for binary packages
06:46.02sylehehe i use to use nfs and rsync
06:46.10QwellThat's no good
06:46.21QwellYou still have to upgrade each machine
06:46.23syleit is if they are all the same
06:46.54sylehell even my password files matched on each server hehe
06:46.58Qwellldap
06:47.07QwellIf you have 100 machines...
06:47.10syleldap is for noobs
06:47.52syleprogram your own shit to do it is better
06:48.02Qwellhis kernels cause him to get disconnected
06:48.43*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
06:48.46sylei;ve never used ldap because i beleive in scalability , if i want to database anything i;ll go directly mysql or postgres
06:48.52Sedoroxspeaking of kernel problems........
06:48.56Qwellpam_mysql?
06:49.04QwellSedorox: <Qwell> his kernels cause him to get disconnected
06:49.13Sedorox:p
06:49.20SedoroxI'm not sure what it is....
06:49.29Sedoroxpast week its been fine.. now its acting up again
06:49.30QwellSedorox: my router does the same thing with the sunhme driver
06:49.35Sedoroxdunno.. probably heat related
06:49.49Qwellthe whole box is freezing?
06:49.50Sedoroxprobably ATI's shitty drivers :p
06:49.55Sedoroxevtually
06:49.57syleoww god
06:49.59syleati :(
06:50.05Sedoroxxchat froze first.. was able to click aim and let people know
06:50.07sylei got a hammer you can use
06:50.10Sedoroxbut then I could click anything else
06:50.11Qwellheh
06:50.31sylenvidia man
06:50.35syleonly way to go
06:50.41Sedoroxyea.. and since 2.6.12 ati's have been broken.. got that fixed with a bad mtrr hack...
06:50.45QwellWhen I bought my ATI, nvidia had zero linux support
06:50.47Sedoroxand yes.. I swear by ATI's now...
06:51.11sylewell they are nice enough to make their own linux drivers
06:51.13syleso that helps
06:51.21Qwellati is too
06:51.22SedoroxLinux Neptune 2.6.14-nitro2 #2 SMP PREEMPT Fri Nov 18 13:05:07 EST 2005 i686 Intel(R) Pentium(R) 4 CPU 3.20GHz GenuineIntel GNU/Linux
06:51.24Sedoroxhmm
06:51.35Sedoroxyea.. but ati's drivers for linux sucks
06:51.39sylemy mythtv box at home runs on my big tv in the living room with no problems
06:51.44Sedoroxand I'm glad to see nvidia has SLI linux support now
06:51.49Qwellnice
06:52.28sylei was thinking i may switch to media center just for better GUI support to my mp3 collection not sure yet, i like ripping dvd's
06:52.47Qwellgui for mp3s?
06:52.48Qwellget a remote
06:52.49sylemy friend gave me a harddrive of about 20 thousand songs
06:52.58Sedoroxdang
06:53.08syleall sorted
06:53.14asterboy20000 songs
06:53.15syleartist album etc
06:53.17asterboy??/
06:53.21Qwelllink?
06:53.22SedoroxI've been thinking about getting a small shuttle machine (or something that small) and making a box for the family living room...
06:53.34Sedoroxrip all the dvd's down.. have my mp3's listed.. rip all their music...
06:53.35syleyeah amazingly enough its not enough, i still missing alot lol
06:53.38Sedoroxwould be nice to have a menu with everything right there
06:53.42Qwellsyle: download link? :p
06:53.49Sedoroxlol
06:53.59sylenot using my bandwidth for that i pay by the gigabyte hehe
06:54.04Qwellaww
06:54.13Qwellhow much per gb? :P
06:54.17Sedoroxlol
06:54.21sylea dollar i think
06:54.27Sedoroxbah
06:54.28Sedoroxcheap
06:54.28Qwelloh, that's nothing
06:54.28Sedorox:p
06:54.38Sedoroxthe box I have colo'd is $8/gig
06:54.40syleit adds up after enough people want it hehe
06:54.56Qwellgive it to one person, and let them use their *cough*my*cough* unlimited bandwidth :P
06:54.57sylei would however be willing to exchange
06:55.03syleyour songs for mine if you got a decent amount
06:55.06sylealways looking for more
06:55.25SedoroxI wanna just goto cogent and get $10/meg on 100mbit commit
06:55.33Qwellouch
06:55.45Qwellthat's freaking expensive
06:55.54Sedorox$10/meg?? no it isn't...
06:56.15Sedoroxat&t wants like.. $32/meg
06:56.15QwellSedorox: I've got a server in TX on a 100mbit link, with 1tb for like $139/mo
06:56.17sylehome box
06:56.19sylelocalhost:~# du -sh /home/warez/mp3
06:56.19syle46G     /home/warez/mp3
06:56.19sylelocalhost:~#
06:56.33SedoroxQwell: well.. thats colo'd.. and monthly limit
06:56.36QwellOnly 17G :(
06:56.37sylei use samba to feed that to rest of comps in the house
06:56.39SedoroxI'm talking straight pipe.. no limits
06:56.50QwellSedorox: but, for $139...you'd only get...14mb?
06:57.01Qwellafter the 1tb, I pay per gb
06:57.11Sedoroxwell no.. would be less.. the $10 is on 100mbit... (maybe gig... I'm not sure...)
06:57.15Sedoroxwould go up for lower commit...
06:57.26QwellMine is on a 100mbit link to a gbit pipe
06:57.33Sedoroxnice
06:57.34Qwellmultigbit, really
06:57.43Sedoroxyea
06:58.07QwellI'm not sure if they do racks, or just leased servers
06:58.15Sedoroxhehe
06:58.25SedoroxI'm looking at eventually getting a box colo'd
06:58.45sylei pay 100 a month for my colo
06:58.46Qwellcolo seems far more expensive than servers
06:58.54Qwellsyle: plus how much for bandwidth?
06:58.55Sedoroxone place I'm looking at is a 1u.. 2mbit (I Think) no limits... for $99/month
06:59.14asterboycolo in your basement for less.
06:59.18Qwellthan dedicated servers*
06:59.27sylehmm i beleive i get 1500 or 2000 GB free with that a month
06:59.32Sedoroxbah.. gotta get a decent pipe in.. and I want multi-homed :p
06:59.42QwellI'm multihomed. :P
06:59.49syleits also one of those huge ass dell 2650's
07:00.17syle8 gigs of ram dual cpu , scsi raid etc
07:00.24Sedoroxpurdy
07:00.31SedoroxI like the pbx we have colo'd.. fast little box...
07:00.48Sedoroxdual PIII 1gig... 512 ecc ram... 18gig U160 (maybe 320) scsi...
07:00.49Sedorox1u..
07:00.53sylei think its the most expensive computer part i ever bought for myself lol
07:00.59Sedoroxcompares to my P4 3.2...
07:01.07syleonly 2 things were more money in my life, my car and my house lol
07:01.22Sedoroxlol
07:01.47Sedoroxyea.. I wanna move my server out of my house... (mainly so I can get Verizon fios when it comes around :p )
07:01.51sylewell house is just a mortgage, so i guess thats leaves just the car lol
07:01.53Sedoroxget it colod...
07:01.54Sedoroxwould be nice...
07:02.06Sedoroxmy car... well.. its old... just maintance on it
07:02.12Sedoroxplus insurance
07:02.23sylei think i paid 13k straight up for my grandam
07:02.33sylenever spend more than that on a car
07:02.34Sedoroxcool
07:02.47SedoroxI got mine as a christmas gift a few years ago :p
07:02.50sylei really want to get rid of it though and get a truck of minivan
07:02.53syleor
07:03.05sylebitch grabbing stuff from home depot etc in a car lol
07:03.18Sedoroxlol
07:03.59sylebut i see people that spend 50k on a car
07:04.09syleand don;t even own a house, crazy
07:04.16Sedoroxyea...
07:04.57syleif you own a 50k car you should own at least 5-10 real estate properties at that point hehe
07:05.00Sedoroxsay.. for linux.. what patchsets to you guys use?
07:05.46sylepatch for what?
07:05.53Sedoroxkernel...
07:06.22syleoww i don;t patch, i just get the new one usually
07:06.27Sedoroxah
07:06.34Sedoroxsome of the patches are nice...
07:06.38Sedoroxbut my main issue...
07:06.40syleafter enough patches i beleive shit can go wrong, so never chose that route
07:06.49SedoroxI need reiser4 support (possibly another reason for my kernel problems...)
07:07.24syledon;t the latest kernels support it?
07:07.32Sedorox.15 might have it in there now...
07:07.34Sedorox.14 doesn't
07:07.57sylewell you;ll have to do it same way you did your other one then
07:08.08SedoroxBTW
07:08.11Sedoroxif you use Reiser4
07:08.15Sedoroxand you use VMWare
07:08.22Sedoroxmake sure you have another partition for VMWare
07:08.23Sedorox:p
07:08.52SedoroxI had to move my /home off R4 because vmware would lock up when creating the virtual disks on a R4 partition for some reason
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07:09.25Sedoroxand of course.. can't use the 1gig low-mem patch either
07:10.07syleyou should always put your biggest partition on its own anyways and format it reiserfs, xfs, whatever is faster for what your doing
07:10.20Sedoroxwell..
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07:10.29Sedorox<PROTECTED>
07:10.39Sedorox( have 2 60gig drives in this)
07:10.50sylethink i;ve tried ext3 and reiser for mythtv but find them to slow on deletes i may try xfs or jfs next
07:10.53Sedoroxsecond had /home and a 20 gig partition for my mp3's for between windows and linux...
07:11.03Sedoroxreiser I like.. ext3.. bah...
07:11.07Sedoroxxfs was nice....
07:11.13Sedoroxuse to use that with slack
07:11.19Sedoroxnver used jfs
07:11.30sylereiser always worked good with running databases for me
07:11.42Sedoroxyea
07:11.46sylehaven;t bothered much with reiser though since ext3
07:11.55Sedorox<PROTECTED>
07:12.00syleany noticeable differences?
07:12.00Sedorox<PROTECTED>
07:12.08SedoroxI haven't really used ext3
07:12.24SedoroxI went from xfs to reiser.. never looked back :p
07:12.30Sedoroxthen decided to try reiser4...
07:12.33Sedoroxstill has its problems
07:12.59sylenoasync is good to if its a mail server and who cares if everything gets deleted
07:12.59Sedorox(one, for example, I can't mount my HDD on the orginal gentoo-patched boot cd that I used to install)
07:13.08Sedoroxlol
07:13.22Sedoroxreiser has saved my ass more then once on that aspect :p
07:14.34sylebut reiser xfs etc etc, don;t replace fast you should be running scsi over ide if you need I/O speed on crucial things like a db :)
07:15.11Sedoroxhmmm
07:15.27Sedoroxfor anything new I build I'll be doing SATA
07:15.31Sedoroxand SATAII for servers
07:15.38syleyeah for home i;d do the same
07:15.44sylejust not work
07:15.52Sedoroxeh
07:15.53Sedoroxdepends
07:16.08Sedoroxif it needs to be a ciritially available server.. the money would be there to go scsi anyway so.. :p
07:16.30sylesata II, hmm i;m behind damn
07:16.36Sedorox3gigabit
07:16.46sylenice
07:16.46Sedoroxclose to being on par with U320
07:17.24sylethey never will be
07:17.30Sedoroxeh
07:17.34sylescsci can multitask ide cannot
07:17.49Sedoroxthat serial scsi looks interesting
07:17.54sylein other words scsi us bidirectional
07:18.01syleis
07:18.30sylei can;t stress how crucial scsi is on a production database hehee
07:18.38Sedoroxyea...
07:19.50sylebecame an expert in perl and its useless with asterisk , can;t tell you how much that pissed me off lol
07:20.06SedoroxI manage one server.. and help with managing another server.. both are scsi systems... 'tis nice...
07:20.14Sedoroxlol
07:20.29sylehad to c code some modules to do what i wanted :(
07:20.37syledamn i hate c
07:20.46SedoroxI kinda liked c++
07:20.51syletake twice as long to do what i could in perl
07:20.51Sedorox<PROTECTED>
07:20.57sylewastes 2 times my life
07:21.31sylefor speed its unmatched
07:21.41syleand thats what asterisk needs
07:21.54sylecan;t be waiting for an interpreter to load before making a call hehe
07:22.08Sedoroxyea
07:22.34syleactually i did one implementation of a fast perl script at first
07:22.38sylefast-agi
07:23.01syleloaded the damn thing in memory first, then when asterisk connected it was already sitting on a socket waiting to respond
07:23.08bsdfreakheh
07:23.39SedoroxI need to update my local box to 1.2.1
07:23.57Sedoroxthen make it like the production box.. work the bugs out. then upgrade the production box...
07:25.51*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
07:25.57sylei prefered working as a sys admin over a programmer at all jobs, i only like programming stuff i wanted hehehe
07:26.12SedoroxI don't like programming
07:26.17SedoroxI like working with hardware
07:26.18sylealthough i usually had to start as a programmer first
07:26.22syleto get into the company
07:26.26Sedoroxespecially networks (any kind.. computer.. telephone... cable... etc...)
07:26.30Sedoroxah
07:27.24syleyeah advertise yourself as a php/perl/c programmer etc, than when you get in, then you can start convicing boss to hire another programmer and make you sys admin as there is so much to do there hehe
07:27.47Sedoroxlol
07:27.55syledoesnt; always work, one time i spent 2 years as a programmer regardless
07:28.01sylelol
07:28.09Sedoroxeheh
07:29.19*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:31.00sylehey man
07:31.07sylehave you looked at the SER source code?
07:31.18sylesat down and read through a bunch of it last night
07:31.48syleit does not look easy to add a module lol
07:32.26SedoroxI haven't messed with SER yet...
07:32.29Sedoroxwant to.. but haven't
07:32.49syleits excellent c code though
07:33.27syleguy who wrote this obviously has at least 10 years c experience or more
07:33.27Sedoroxlol
07:34.05sylewell i been looking at a simple load balancing ability
07:35.24syleSER seems it would be good if i can break the source code to understand it, but then your missing iax and h323 support, so i was hoping there was something like SER for asterisk
07:36.50sylethat would probaly be pretty popular some third party deamon doing redirects for sip, iax, and h323
07:38.18*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
07:38.42syleor do what everyone else does, fuckyou just use sip
07:39.03sylellol
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07:42.54Vendetta9does anyone use a D-Link DVG-1120s router with asterisk?
07:43.33QwellIs that the one that screws up sip?
07:44.01Sedoroxlol
07:44.10Qwellno, seriously
07:44.24Vendetta9don't know.  was an mgcp device that untilt he firmware was changed to allow sip
07:44.32Qwelloh, it's a voip router?
07:44.41Vendetta9yea
07:45.30Vendetta9been searching the net to find if anyone has been able to use it with * but little success
07:46.28sylereading up sedorox keep your / ext3 and make biggest partition resierfs, best way to go
07:46.40Sedoroxhmm
07:46.48Sedoroxwell I guess my / would be my biggest partition :p
07:47.02syleif you need anymore space in / anywhere just symlink whatever you need
07:47.15Sedoroxhehe
07:47.28Sedorox<PROTECTED>
07:47.35Sedorox<PROTECTED>
07:47.40Sedoroxdamn I use a lot of space
07:47.50Qwellpfft
07:47.54sylelol
07:47.59Qwellwhat's that, 10gb?
07:48.01sylebuy some 300 gig drives
07:48.11SedoroxI have a 200gig in my one server here
07:48.14Sedoroxgot my mp3's on it now
07:48.25Sedoroxboth these drives are 60gig
07:48.37Sedoroxits a laptop.. so its kinda pricey to upgade to 100 or more
07:48.52sylelol
07:48.55Qwell# du -hs /mnt/storage/pr0n/
07:48.55Qwell76G     /mnt/storage/pr0n/
07:49.04Sedoroxlol
07:49.11syleyou don't use laptops to build storage solutions :)
07:49.12SedoroxI have 11gig of video sittin gin /root
07:49.29Sedoroxfrom my grad project
07:49.44sylei am waiting till motherboards support 5+ sata connectors
07:49.51sylegonna raid 5 the bitches
07:50.02Sedoroxlol
07:50.18syleall 300 gig drives
07:50.24SedoroxI was gonna do raid0+reiser4 to have a quick fs...
07:50.25SedoroxAHAHHA
07:50.29sylegimme a terrabyte or so for all my movies etc
07:50.32Sedoroxthen I realised.. its software raid
07:50.35Sedoroxwas like screw that
07:50.52Sedoroxbut at least they do offer raid 0,1 from the controller (promose fast track)
07:51.02Qwellsyle: just build a san
07:51.02syleyeah well software raid is only way to go for a home system
07:51.28sylenot going to go spend thousand bucks on a hardware raid controller when linux can do it hehe
07:51.29Sedoroxmeh
07:51.39Sedoroxsata is hardware raid by standard I think
07:51.51Qwellnah
07:52.04Sedoroxhmmm
07:52.19syleyes that bothers me
07:52.28sylei need someway for the kernel to report a failed disk
07:52.29SedoroxI thought it was.. huh
07:52.34sylebefore another one goes and all my data
07:53.28Sedoroxhmm
07:53.30sylenew system will come with raid 5 in the future
07:53.39syleand lcd screen to tell you this
07:53.58sylebut doesn;t help designing a program to email you if their is a failed disk hehe
07:54.10Sedoroxlol
07:54.28Sedoroxdamn.. -mm is huge...
07:54.56sylewell my idea is to setup a basic system at home, 5 drives, raid 5 in the kernel, get it all setup, pull some power connectors off some drives, see how the kernel responds, etc
07:55.39sylewrite some simple perl script to email me if their is a failed drive
07:55.41sylebout it
07:56.47sylei have about 200 movies
07:57.03syleand its a pain in the ass to try and find a certain movie
07:57.20sylei want to call it up on mythtv and just play it from my remote lol
07:57.47Sedoroxlol
07:58.43sylei may move to media center so i;m hoping i can somehow mount a samba share, but knowing MS they probably have some custom movie format to play movies screwing everything up again
08:00.13sylei beleive the only thing security accomplishes is a way for someone to program/make something to make money by disabling that security
08:00.29sylexbox modchips, movie copiers etc etc
08:00.53Sedoroxhmm
08:01.23sylebe the first to mod the xbox360 and put out a modchip in next month, you;ll be a millionare in a year
08:01.30syleetc
08:02.16Qwellyeah, because everybody else who made modchips for every other console are rich now :P
08:02.46QwellWhat you need to do, is invoke the dmca when they steal your design. :D
08:02.58sylelol
08:03.12Sedoroxlol
08:03.36bsdfreakheh
08:06.09*** join/#asterisk xianlp (n=xian_1@193.170.41.114)
08:06.22xianlpneed help
08:06.26sylethink anyone is going to code that skype to asterisk module
08:06.47sylethat bounty has been there forever lol, up to 1500 US now
08:06.54xianlpis somebody here who knows if i can make my asterisk phone one of my phones at a specified time?
08:08.14*** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it)
08:08.16sylegotoiftime
08:09.06Qwellcall phone
08:09.09Qwellerm
08:09.10Qwellcall file
08:09.18Qwellhe'll know the answer. :)
08:10.08syleowww phone
08:10.21sylewrite a perl script to generate a callfile and throw it in cron
08:11.29xianlpthx i'll try
08:12.44Sedoroxreal    7m3.688s
08:12.44Sedoroxuser    10m46.764s
08:12.44Sedoroxsys     2m8.833s
08:12.47Sedoroxmake -j3 for kernel
08:13.04SedoroxP4 3.2 HT/EE.. 1gig ram
08:13.54*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:14.02sylemy question to you is
08:14.12sylewhy does a single processor machien need a gig of ram :)
08:14.48Sedoroxwell its HT.. so its 2 according to the OS.. but yea... we all know ht sucks .. :p
08:14.57Sedoroxwell.. its my primary machine.. I push a lot through it
08:15.07Sedoroxbe it gaming.. compiling (gentoo)
08:15.15Sedoroxand some other testing stuff I do...
08:15.47syleyou know i was expecting the new game consoles to run at 3.2ghz
08:15.59syleunder 900mhz
08:16.03Sedoroxhmmm
08:16.06syleso they only doubled
08:16.30Sedoroxisn't 360 tri-core tho?
08:16.46coppicea gig of ram is nothing these days. do any media work and you soon want more
08:17.15syleits linux not some XP machine running miya
08:17.29Sedoroxlol
08:17.30Sedoroxwell yea
08:17.37SedoroxPremier runs very nice with a gig of ram :p
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08:18.28sylei;ve never needed a machine with more than 512 megs of ram for photoshop etc
08:18.44Sedoroxwhen your running realtime video editing... the more ram.. the better
08:18.49syleso even for most people as a client it seems a waste
08:19.02Sedoroxin fact.. over break.. I'll be helping my old boss build the new editing station....
08:19.06coppiceeven for audio, you soon want more than a gig
08:19.08SedoroxAthlon 64 X2
08:19.17SedoroxI think a gig or more of ram
08:19.27Sedoroxfew 100+ gig sata drives
08:19.33syleyeah if your dealing with video for sure
08:19.48Sedoroxyea
08:19.59sylewhat is the typical buffer cache these days
08:20.10Sedoroxespecially since this box will be used for live switching enviroment/video passtrhough adding texgt and stuff
08:20.14Sedoroxon drives?
08:20.22Sedorox8meg I think... 16 is becoming more common
08:20.51Sedoroxbrb.. kernel upgrade.. hopefully fglrx will load....
08:20.56Sedorox'tho it doesn't look like it will
08:21.07sylehmmm i don;t know much about video, when you buffer an audio stream off the internet is that buffer saved to disk or memory
08:21.16*** join/#asterisk bakermd (n=mbaker@exchange.i2telecom.com)
08:21.22syleprob disk
08:21.29sylefuture will be ram
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08:24.25bakermdHey folks, I am getting the issue "Inband DTMF is not supported on codec g729. Use RFC2833" - but I have dtmfmode rfc2833 in my sip.conf
08:25.06sylein sip.conf or in your context your getting that
08:25.44bakermdthe error?  I get that in asterisk -r
08:25.48syletry adding it to your context first to make sure your other contexts aren;t changing it
08:26.08syleno inband=
08:26.15bakermdok
08:26.53syleerr dtmfmode
08:26.54bakermdI am not sure I know how to do that
08:27.37syleedit sip.conf
08:27.43syleedit context that is saying that
08:27.53syleadd dtmfmode=inband for example
08:28.12syleunder host secret etc
08:28.52bakermdokay
08:29.03syleand you;d only be changing dtmfmode if your carrier is using that so they should be giving you this info
08:29.10bakermdI have 4 total contexts - General and 3 different servers
08:29.29bakermdAll of our carriers support it
08:29.39*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
08:29.45syleowww
08:29.45Sedoroxah.. was right.. ati broken yet again
08:29.50syleyour using g729
08:29.53bakermdall are context inbound
08:29.54sylechange code to ulaw
08:29.57sylecodec
08:30.02Sedoroxg729 is nice!
08:30.03sylethat will fix it
08:30.05bakermddtmfmode rfc2833
08:30.24bakermdreally
08:30.30bakermdWhat is going on?
08:30.51bakermdulaw is 711 - the carrier does not support 711
08:31.40syleummm
08:31.43syleyour not reading
08:31.47syledtmfmode=info
08:31.51syleis rfc2833
08:32.20syleerr maybe not
08:32.37bakermdI thought info and rfc2833 were competetors
08:33.09*** join/#asterisk Ansonmus (n=ahaeser@dsl97-13-100.fastxdsl.nl)
08:33.15syledtmfmode = rfc2833
08:33.18syleyou have that?
08:33.20bakermdYes
08:33.27bakermdIn each context
08:33.31syleyou do a sip reload?
08:33.33bakermdYes
08:34.15sylewhere you get your g729 from?
08:34.20bakermdDigium
08:34.44sylewell your message states you have something like dtmfmode=inband somewhere in that file
08:34.46syledid you search?
08:35.07SedoroxNeptune Grad Project # du -sh
08:35.07Sedorox11G     .
08:35.11Sedoroxyay for nothing but video
08:35.45bakermdthe file is real short - not a big config here
08:36.18sylethats strange it should default to rfc2833 yet you default to inband
08:36.50bakermdIt's a config in DB, not file, but still - changes take effect on a sip reload
08:36.52bakermd<PROTECTED>
08:37.09sylegeneral should never be put in db
08:37.27syleno reason for it, you don;t edit that
08:37.28*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
08:37.54bakermdWell, I just added that as a troubleshooting measure
08:38.03bakermdI will remove it
08:38.46bakermdall general is gone
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08:39.14syleworks now?
08:39.19bakermdno
08:39.45sylesearch sip.conf sip.conf.ext and your whole db for dtmfmode and check the values
08:39.53bakermdokay
08:41.39syleif you still can;t find anything remove sip realtime from extconfig.conf and try again, isolate the problem
08:42.49sylebut those 3 contexts you have in sip.conf you should be able to add dtmfmode=whatever to every single one of them, thats last thing to try
08:45.06bakermdYeah, dtmfmode rfc2833 is set for each of the contexts.. there are no files using inband (other than mgcp.conf, and we have no mgcp going on)
08:47.09syleyou sure asterisk -r is restarting it
08:47.25sylehave you tried shutting down completely and restarting
08:47.59bakermdI will try that tommorrow, when I can have more downtime.
08:48.03bakermdThanks for your help
08:48.11syleno downtime
08:48.13syleasterisk -r
08:48.15sylesip reload
08:48.33bakermdasterisk -r just gets me into the monitor
08:48.41bakermdthen I do a "sip reload"
08:48.56syleare you sure you didn;t do something stupid like set rtcachefriends to yes or leave it at default
08:49.39bakermdleave it at default is a possibility
08:49.42bakermdwhere is that set?
08:49.48sylethat would be if your changing sip entries all the time
08:49.54sylein the db
08:50.05syleits in general
08:51.08bakermdI have no entries for anything remotely similar to rtcachefriends
08:51.13sylesip reload should have caught your changes though but i don;t know enough about how the realtime driver reacts on that command
08:51.46syleumm its documented in the example sip.conf's
08:51.50bakermdokay
08:53.50syleif you don;t set that to no you'll have to reload sip config file all the time, not much point in realtime then
08:54.31bakermdFriends do not change for me though - have had the same 4 "friends" for months, and will not be changing them for a while
08:55.14bakermdI mean, other than adding this codec, I have never even touched this * box, and the guy that set it up left the company 2 months ago
08:55.19sylei;d take sip realtime right out then
08:55.44syleis your codec working?
08:55.48syleshow translations
08:56.02sylewithout the s
08:56.30bakermd<PROTECTED>
08:56.39syleok its working
08:57.10sylebest i can suggest is 2 things
08:57.16*** join/#asterisk lorinc (n=ang@caracas-3384.adsl.interware.hu)
08:57.32syleget example sip.conf and take it out of realtime enabling mode you want to isolate the problem
08:57.42syleand completely restart asterisk
08:57.46bakermdokay
08:57.56bakermdWill do.  Thank you for your assistance
08:58.03syleif that don;t work then i;d start to look more closely at your codec
08:58.06sylebut gl
08:58.10bakermdThanks!
08:58.12*** part/#asterisk bakermd (n=mbaker@exchange.i2telecom.com)
09:18.55Sedoroxnight
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09:26.11mkl1525Hi, I've got an asterisk with an ISDN card that should work as voicemailbox for all the other ISDN phones. When the person on the ISDN-phone activates call forwarding to the number of the asterisk I get the caller id of the incoming call but is there a way to get the number the call was original intended to? Or is there a better setup for such a solution?
09:28.24*** join/#asterisk Dutts (n=dutts@81.168.70.41)
09:29.13DuttsHi, Can anyone help me, I upgraded last night to 1.2.1, downloaded the source, comiled it for libpri, zaptel and asterisk, did make install, now I get Loding module chan_zap.so failed!
09:35.28benjkdid you reboot?
09:36.33*** join/#asterisk sofh (n=ok@203.101.180.165)
09:40.57QwellDutts: You may need to reload the zaptel drivers
09:51.32benjksome people here seem to ask questions but disappear if you don't respond to them within 20 seconds
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11:06.09Duttsbenjk... sorry I had to nip out.... wasn't because no-one replied =)
11:06.22DuttsQwell: so I need to download and comile them then?
11:07.15benjkDutts: np
11:07.36benjkyou will need to make sure that your new zaptel drivers will be loaded into the kernel
11:07.49benjkcompiling alone won't do that for you
11:10.22Duttsaha..... ok so where can I get the latest zaptel drivers? I got zaptel form the digium cvs when I grabbed libpri and asterisk and comiled and make install'd them all
11:11.13benjkyou should try to get all modules with the same version number/released date, if available
11:11.35Duttsok I'm going to try and grab them again from cvs and compile, I've removed all the old stuff from /usr/src and I'll grab them again
11:11.59benjkthen if you recompile them all, and do make install, you will get the new Asterisk once you restart, but you will still have the previous zaptel drivers in the kernel
11:12.30benjkthe easiest thing would be to reboot so that upon system restart the newly compiled drivers will be loaded
11:12.42Duttsaha I see..... so how do I get he latest zaptel drivers into the kernel?
11:14.24Duttsalso is the digium cvs down? I get  cannot write to history file /usr/cvsroot/CVSROOT/history: Permission denied
11:15.20*** join/#asterisk Poincare (n=jefffnod@195.207.137.89)
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11:22.18Duttsanyone else having a problem with the digium cvs?
11:23.42benjkas I said, the easiest thing is to reboot
11:23.59benjkif you have done "make install" in zaptel and libpri before
11:24.13DuttsI have already rebooted, but I'll try again
11:25.11*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
11:26.03Duttsfirst off I logged into the cvs, downlaoded the 1.2 src for libpri, zaptel and asterisk, then I did a make clean, make, make install on all of them. make on asterisk warned me of some modules which were incompatible, so I google'd the error and found that I had to delete the modules dir, so I did that, did the whole thign again and then got the failed to load chan_zap error. So I deleted all the src and went to downlaod it again but now the cvs is not letting
11:26.22*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
11:26.44Duttsthroughout this I rebooted also but I'm trying again now
11:30.47Duttsok just rebooted, and I still got the same problem... anythign else I can do?
11:30.47DuttsI get this first Dec 10 11:06:27 WARNING[800]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device or address
11:30.47benjkdid you build Asterisk before or after Zaptel?
11:30.47Duttslibpri, zaptel, asterisk in that order
11:30.55Duttson a system that worked for for about 12 months =(
11:31.01benjkthat's a different error though
11:31.16Duttsone sec I'll pastebin the whole thing
11:31.26benjkdon't need it
11:31.54benjktry to run ztcfg and see if you get any feeback
11:32.30DuttsI don't get anything back.....
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11:40.33benjkdid the driver actually load?
11:40.38benjklsmod
11:41.46Duttsyes it did.... now asterisk works?!?!?!? I never used to have to run ztcfg before asterisk? is this new?
11:42.16Duttsjust looking at my rc.local and it did modprobe zaptel and wcfxs and then started asterisk
11:43.08benjkno, usually Asterisk will do what ztcfg does
11:44.26Duttshmmm I've just tried a reboot again, and asterisk wont work unless I run ztcfg first?!?!?!? also I don't lways get the cli from asterisk.... I'm starting to not liek 1.2 at all =(
11:45.09benjksounds like Asterisk didn't find the configs for Zaptel
11:46.48Duttsaha... coul dit be because I used to run as non-root..... then I deleted the modified makefile when I recompiled my source, could that break it now?
11:47.03benjkcould be
11:47.46*** join/#asterisk k31th (n=Kevin@flashtek-uk.com)
11:48.02Duttswould that cause asterisk to not display a cli when I do 'asterisk' then 'asterisk -r' ?
11:48.07benjkas I said, it looks like ztcfg finds the config and Asterisk doesn't, could be an access problem
11:48.18k31thasterisk -vvvvc
11:48.34k31thi belive -c is the cli ?
11:48.41k31thv is for verbose
11:48.42Duttseven when you issue -r?
11:48.43benjkc is for console
11:48.51benjkto start in console mode
11:48.56benjkas opposed to background
11:48.58k31thahh
11:49.00k31thyea
11:49.08k31thnot sure wat the -r switch does tbg
11:49.09k31thtbh
11:49.12benjkr connects to a background asterisk process
11:49.18k31thahh
11:49.25k31thi always like to see wats going on
11:49.32k31thso i run it in a screen
11:49.51benjkwell, if you run Asterisk in console mode, it will be tied to your terminal session
11:50.01benjkthat's ok for trouble shooting
11:50.08benjkbut not such a good idea for a server
11:50.21Duttswhat is the default value of ASTVARRUNDIR ?
11:50.26Duttsin asterisk.conf
11:50.34benjkif you run a server that people have to rely on you need to run Asterisk in the background
11:51.04benjkshould be /var/run
11:51.12k31thbenjk: just dc the screen
11:51.13k31th:D
11:51.32benjkif you run Asterisk as non-root, it would have to have its own directory in /var/run
11:51.40benjkfor example /var/run/asterisk
11:51.45k31thatm im only testing asterisk
11:51.58k31thwen its all working ill -r it
11:52.36benjk-r is only a remote console that connects to an already running asterisk background process
11:52.43k31thoh
11:52.50k31thwell ill run it in back ground
11:53.01benjkyou start asterisk as a background process by doing /usr/sbin/asterisk
11:53.05k31thi need to get voicemail working first :p
11:53.07benjkwithout any arguments
11:53.17k31thyeah i just type asterisk
11:53.22k31th:D
11:53.37k31thwat distro do you guys run asterisk on ?
11:53.37benjkthen you can connect a console to that background process doing /usr/sbin/asterisk -r
11:53.49k31thsweet
11:54.06benjkbut /usr/sbin/asterisk -c will start asterisk in the foreground, attached to your terminal
11:54.10k31thatm i have it running on debian and gentoo
11:54.35Duttsthanks for your help guys I think it mightbe my non-root stuff screwing things up. I'm goign to go back as runnign udner root, but I need to be able to download the src and recompile again.... does anyone know if the cvs is down as I cannot seem to get any code?
11:55.04k31thDutts: does your distro not have a package or a ports system ?
11:55.16k31thif not u could always grab it from there ftp
11:55.25k31thunless u wanted current
11:55.28DuttsI'm uding RH8
11:55.35k31thwhy ?
11:55.38k31ththats yrs old
11:56.12Duttsyes I know..... dunno to be honest, just the box I had lying around.... downloaded FC4 CDs this morning so migth install that and go from there instead
11:56.36k31thhum yeah we use centos at work
11:56.41k31thnot a fan of it much tbh
11:56.51k31thits good that it has RHE binary compatability
11:57.09k31thbut iv never needed it, i prefer debian for a server better pkg man
11:57.10Duttsso what distros u guys use then?
11:57.16Duttsah debian
11:57.20k31thdebian, gentoo
11:57.33k31thDebian Sarge works nice
11:57.49k31thDutts: do the net install and then install nothing
11:58.10Duttsnoth8ng?
11:58.11k31thboot up login as root apt-get update; apt-get install asterisk
11:58.21k31thna i just leave the base system
11:58.24Duttsoic
11:58.31k31thunless u need any thing else
11:58.39k31thi then install wat i want with apt
11:58.47Duttsnoep this is a dedicated box, only need text mode install also as it's quite low spec.... P2 400
11:58.54k31thyeah
11:58.58k31thdo that then
11:59.08k31thu can have it working in an hour easy :D
11:59.19Duttssweet I';ll give that a go.... cheers for you help
11:59.51k31thnet install iso is about 140mb
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12:33.26foo8arjust upgraded to 1.2.0 and my uniden reports unavailable :(
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12:36.22foo8arthe uip200 doesn't reply to * polls
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12:51.10kredfordyou mean no one has nothing to say
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13:02.58Teelii have an ATA from a eusso. when i try to dial to it, its console says RTP port not opened. i can see port number in SIP headers
13:03.21Teeliwhy would that happen?
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13:14.53Teeliwhen does asterisk open RTP port for an outbound call. does it open at the time when it makes call or waits for ACK before opening port
13:23.15nassyis a GSM gateway useful for a regular company. ie, one that is not selling cellular service
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13:26.39nassynevermind i found my answer
13:27.16gnosysAnyone here tried out Digium's new card: the TDM2400P with new VoiceBus Technology.
13:27.19gnosys?
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13:35.33gnosyshi [tk]d-fender
13:36.00[TK]D-Fendery0
13:36.22gnosysHave you done any fiddling with Digium's new TDM2400P?
13:37.19[TK]D-Fendernope, I run a Rhino Channel bank at work and no need at home...
13:37.30[TK]D-FenderWhat kind of situation are you considering it for?
13:38.19gnosysWell, I'm not seriously considering buying one for myself just now... more curious about the new VoiceBus Technology.  How does one connect the analog telephone lines or the analog telephones to the card.
13:40.12[TK]D-FenderIt has an amphenol connector out the back which is standard telco termination (normlaly yuo get a cable that leads to a punch-down block like BIX) or you buy a "break-out" box like this : http://www.canadianvoipstore.com/index.php?cPath=99_300_305
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13:40.45[TK]D-Fenderhow many lines / extensions?
13:41.36gnosysyou asking me?  how many lines/extensions i need?  not as many as that card has.  I'm just curious about it.  I'll read up at your pointer.  thanks!
13:43.05[TK]D-FenderOk, business, personal, or just curious?
13:43.11gnosysnow i understand better.
13:44.15gnosysI'm curious about it for business purposes.  Just exploring the capabilities is all.  I'm not sure when they released it, but before today when I noticed it, I thought that I was limited to 4 analog ports per card and this really changes the picture quite alot!
13:44.43foo8army uip200s doesn't respond to sip keepalive after upgrading to 1.2.0. could i disable the keepalive so that the destinations doesn't get unavailable?
13:46.49gnosysHow many lines in your Rhino channel bank?  Do you like that product?  Does it work with *?
13:46.53tengulrehi,all! I m coming!!! :)
13:47.23tengulrenice to meet everyone!!
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13:48.21[TK]D-FenderI use it for 4 FXO (backup POTS) and 8 FXS (for fax machines).  It does work, but my fax reliability is pretty chaky.  It plugs into my 4 port T1 card.  Wouldn't suggest it really...
13:48.25gnosyshi tenguire... sounds like you're having sex or something!!!
13:48.36[TK]D-FenderHow many lines are you envisioning using?
13:48.52tengulrey?
13:52.11gnosysWell, (long-term planning mode) I can envision setting up an * PBX for a mid-sized enterprise with 24 analog lines and 24 extensions (or 12 analog lines and 36 extensions), but that's just pie-in-the-sky right now.  I need to learn more about configuring my TDM04B, TDM40B, and my old X100P (I think that's what it's called).
13:52.39gnosysBefore I go far with the pie in the sky
13:55.39[TK]D-Fendergnosys : Forget analog "lines"  Get a T1 PRI.  No kidding.  and FORGET about the TDM2400 in that case.  For your analog phones (lord knows why you'd want to if you didn't have to) I'd suggest you use ATA's
13:55.50[TK]D-FenderFor that QTY of lines ayways
13:58.30gnosysyou misunderstand...  I wasn't thinking of intentionally putting together something like that, preferentially, over a T1.  I was thinking if I ran across a client who already had that many analog POTS lines and/or that many extensions, how I could hook up the client with an * PBX.  That's all.  I just noticed the new product and was curious about how and where it might best be used.  It seems reasonable to think that there is a market for it
14:01.41[TK]D-FenderYeah, older companies who'd have problems rewiring for VoIP and don't use digital set like Norstar/Avaya/Lucent taht can be converted.  Though I would still never suggest that card in place of SIP gateways.
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14:02.20burnproofgood day guys
14:02.33burnproofanyone here try video support on asterisk ?
14:02.49burnproofi'm having a little problem with regards to codec
14:03.07burnproofmy softphone is eyebeam care to share TIA
14:06.34gnosysWhere can I read about SIP gateways, [TK]D-Fender?
14:08.24foo8army uip200s doesn't respond to sip keepalive after upgrading to 1.2.0. could i disable the keepalive so that the destinations doesn't get unavailable?
14:08.46tengulrehi,all! where have SIP source code for windows client?
14:08.58[TK]D-FenderWell not sure what there is to say about them... check out www.voipsupply.com for pricing.  Sipura's are the best low density (1-2ports) for which you might use multiple, and in the 4-8 port range things get much more expensive per port, then cheaper as you hit 24.
14:09.26[TK]D-Fenderfoo8ar : Use "nat=never" in SIP.conf
14:09.53[TK]D-FenderAnd "qualify=no"
14:10.46foo8ard-fender: did that, but get "chan_sip.c:11396 sip_poke_noanswer: Peer '9004' is now UNREACHABLE!  Last qualify: 0" directly after startup
14:10.51foo8araha
14:10.54foo8arsorry
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14:12.23foo8ardid the trick THX!
14:12.58[TK]D-Fender:)
14:13.05foo8arwho is wrong, the UIP or * SIP implementation?
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14:13.10[TK]D-FenderI hate my 2 UIP-200's :/
14:13.18[TK]D-FenderUIP = bleh
14:13.57markithi :) is it possible that after the last checout of 1.2stable, my sip budgetone can't be registered anymore? I get this error: WARNING[30019]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0x81b5068 (len 533) to 192.168.1.102:5060 returned -1: Operation not permitted
14:14.00[TK]D-FenderI thought they'd be nice, but only for 1 thing : Cheap PoE phone for my company's entranceway which is a beigger target for VANDALISM. (and it wall-mounts easy)
14:14.12markitwhen I tried 1.2 the first time, it worked fine
14:14.55gnosysSo, [TK]D-Fender, from thinking about the phrase "SIP Gateway" and checking out the products as voipsupply.com, and thinking about what we were talking about, I suppose that a SIP Gateway is designed to let you connect analog lines into a VoIP network for outbound connectivity (rather than having a T1 be the outbound line)?  Or perhaps, to let you connect analog phones into a VoIP network?  Still learning here...
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14:16.54gnosys[TK]D-Fender, you mentioned that you have a SPA-941 yesterday and that you were very happy with it.  What is the mfgr's policy on firmware and config files and docs?  Like Cisco or Polycom? (or other?)
14:17.01[TK]D-FenderEither.  ATA's typically refers to pluuing analog PHONES in and putting out SIP.  GATEWAYS typically imply taking analog LINES in and spitting out SIP.
14:17.26gnosysok.
14:17.27[TK]D-FenderYou can get them in all sorts or combinations (Sipura SPA-3000 = 1 FXS, 1 FXO), and certain others
14:17.54[TK]D-FenderUSUALLY the larger ones will ball "all of one kind" only.
14:18.16[TK]D-Fenderfor either all FXS or all FXO
14:20.16gnosysI'm learning alot in this conversation with you about stuff I didn't know much about before.  What would you suggest as good reading material that would tell me about all of the different components in this picture?  Seems as though I've only learned about a few of the many different kinds of "Legos" in the analog/VoIP telephony world.
14:21.29znoGdoes anyone use distinctive ring with Asterisk?
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14:22.31[TK]D-FenderI'd say read up on T1 tech (PRI), and channel banks and that covers most oher bits aside from analog->SIP converters
14:24.40markithow can I test if a sip packet can reach my phone? ping works, but for a sip try?
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14:32.45[TK]D-FenderOh... about the Sipura stuff : Not sure on returns (believe their pretty good though), and for firmware, there are relatively regular releases and is pretty stable  Doc's are a little technical, but fairly complete if cryptic at times.
14:33.08[TK]D-FenderBut I've never run across one I couldn't get to do what I wanted in a few minutes :)
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14:45.38znoGdoes anyone use distinctive ring detection with Asterisk?
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14:54.25ManxPowerznoG, I don't think many people do.
14:57.05znoGManxPower: I agree, but I figured someone must.
14:57.18ManxPowerznoG, check the mailinglist archive.
14:57.28znoGManxPower: i've been doing just that
14:58.02znoGManxPower: i just want to find one set of config files from someone who has set it up with distinctive ring detection
14:58.16RoyK"Still have a callno..."
14:58.18RoyKwtf?
14:58.22ManxPowerznoG, the problem is that it's done differently in different countries.
14:58.23RoyKand server spinning on 100% cpu
14:58.26RoyKand not accepting calls
14:58.29RoyK:|*
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15:01.22znoGManxPower: irrespective of which country one is in, the problem I'm having is that as soon as the call comes in, Asterisk doesn't even wait 2 seconds to determine the type of ring cadence and tries to go to pattern 0,0,0 which doesn't exist.
15:01.35znoGManxPower: i don't know if I should have a dring pattern configured for 0,0,0
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15:03.44pigpen2anyone around...I have a little battle this morning with Asterisk...?
15:04.17pigpen2I am issuing the dial command to several sip extentions..
15:04.26pigpen2It works great with snom
15:05.21pigpen2But I just deployed 3 Polycom 601's.  When the call is sent to about 10 sip phones, and someone answers....the polycom's still ring...
15:05.35pigpen2The users actually have to answer the phone and hang up.
15:05.51pigpen2Quick ideas would be great...as I am on site.
15:07.29[TK]D-Fenderpigpen2 : What SIP version?
15:07.31jake1932pigpen2: did you do a sip debug?
15:07.53pigpen2Polycom Sip 1.6.2 I think.
15:08.03[TK]D-FenderAnd pastebin your dialing section extensions.conf
15:08.20pigpen2not yet...should I look for the cancel command??
15:09.00foo8arhow could you implement callback with possibility to dialout?
15:09.31pigpen2Well, the relevant lines are:
15:09.32pigpen2exten => s,1,Dial(SIP/100,15)
15:09.32pigpen2#exten => s,2,Dial(SIP/100&SIP/112,15)
15:09.32pigpen2exten => s,2,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111,15)
15:09.35ManxPowerpigpen2, I've not seen that issue with my polycoms
15:09.43pigpen2tks Manx
15:09.57[TK]D-Fenderhmmm, same here
15:09.57ManxPowerpigpen2, with that many devices to fing, consider a SIMPLE queue.
15:09.58pigpen2It worked great before the new 601's
15:10.12pigpen2hmm...I haven't done that...
15:10.22ManxPowerpigpen2, all the docs make it look complicated, it's not.
15:10.42pigpen2Yeah...that is what I was thinking...
15:10.53jake1932ManXPower - that goes against logic
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15:11.02ManxPowerjake1932, Why?
15:11.09jake1932all the docs make it look complicated - it's probably complicated
15:11.18pigpen2So after the receptionist doesn't answer, it would fall to the queue, the queue would ring the queue members?
15:12.45jake1932pigpen2: i know from my phones, i can check to see they receive the cancel cmd.  not sure if you can do that with the polycoms
15:13.04ManxPowerhttp://pastebin.ca/33183
15:13.09ManxPowerjake1932, It CAN be complicated.
15:13.36pigpen2ManxPower, thanks.
15:13.39ManxPowerSpecifically the Agents stuff is complicated, but I don't use agents.
15:13.59pigpen2ManxPower, you have helped me several times...thanks again.
15:14.08ManxPowerpigpen2, you're welcome.
15:15.30pigpen2Ok..I am going to go give it a try...bbiab.
15:16.39[TK]D-FenderManxPower : Why 6 calls to the same phone?
15:16.53ManxPower[TK]D-Fender, because the phone only has 6 lines.
15:17.35[TK]D-FenderHave you considered using 1 reg with 6 lines keys to it?  A LOT simpler
15:18.10[TK]D-Fendertahts what I do with mine, though I'm dropping mine to 3 line-keys so users have room for buddy-watch / speed-dials
15:18.13ManxPower[TK]D-Fender, yes, but in the past that caused all sorts of problems.
15:18.18[TK]D-Fenderwhat kind?
15:18.29ManxPower[TK]D-Fender, not being able to transfer a call.
15:18.47ManxPowerAlso it breaks our "model" which is "one account per reg".
15:18.57ManxPowerAlso it breaks our "model" which is "one account/reg per line"
15:19.12ManxPowerMany of our users want different extensions on different lines
15:19.22[TK]D-FenderNever had that problem here.... what happened?
15:19.56ManxPower[TK]D-Fender, Ah, no, the problem was if all lines on the phone only have 1 reg, then the user could NOT conference calls.
15:20.33ManxPowerAlso how would I direct calls to the 2nd or 3rd line on a phone if I did that?
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15:21.28ManxPowersome of our phones are Line 1: Outgoing calls only, Line 2: personal line, Line 3 & 4: the person's boss's first 2 lines.
15:21.40[TK]D-FenderWell using 1 reg w/ 6 key's, subsequent incoming/outgoing calls would just use the next free line key.  I've conferenced in 3 people at a time with it without problem.  as for "lines" since I use 1 reg its 1 extension = 6 simultaneous calls.
15:21.59ManxPower[TK]D-Fender, I'm sure the big was fixed in the past year.
15:22.24ManxPower[TK]D-Fender, most of our phones have more than one extension on them.
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15:25.13ManxPowerWe have 1 user that seems to go thru 1 assistant per month.  I finally stopped updating the callerid for that phone and just set it to "Temp Assistant"
15:27.22ManxPowerOur users are Real Estate agents.  They think the world revolves around them.
15:28.42[TK]D-Fenderhehe. I've been al god since I got my 600's with SIP 1.5.2 & BR 2.6.1 in May
15:28.47[TK]D-Fenderall good*
15:29.10robl^try working with lawyers!    they are worse than real estate agents...
15:29.44[TK]D-FenderIs there a secondary bonus for more than 1 extension/phone in yur scenario?
15:29.44ManxPower[TK]D-Fender, being able to direct a call to a specific line.
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15:31.11[TK]D-FenderBut as you could treat an incoming call as just an incoming call, what does targeting a specific line key do for you?
15:32.17ManxPower[TK]D-Fender, it allows us to have different extensions on the same phone and the call comes in on the line with that extension on the label.
15:32.52ManxPowerWe have many people that share the SAME phone, each person has their own extension on their own line appearance.
15:32.57blitzragemorning
15:33.04[TK]D-FenderSo your workers are very task oriented and for seperation purposes find that the way to go?
15:33.12[TK]D-FenderOHO!  Shared phones?
15:33.25[TK]D-Fenderblitzrage : Got yours up & running?
15:33.33[TK]D-FenderEEK
15:33.34ManxPowerWe also have many phones in "boss and assistant" setup.
15:33.40blitzragenope... not up :)
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15:33.45[TK]D-FenderOk, wierd :)
15:33.59[TK]D-Fenderblitzrage : Due to your focus, or failure?
15:34.10[TK]D-FenderShould work "out of the box"
15:34.21blitzrage[TK]D-Fender: due to lack of PBX to hook the phone up to -- we tore down the class yesterday
15:34.33[TK]D-Fenderhehe
15:34.37[TK]D-Fenderoh well!
15:34.47blitzrageyah -- thats what happens when you're on the road
15:34.50[TK]D-FenderNo extra accounts?
15:35.08blitzragewhat do you mean?
15:35.13blitzrageI don't have anything to plug the phone into
15:35.15[TK]D-Fenderthough the 2 city ones you told me of were fixed
15:35.25[TK]D-FenderYou mean no network?
15:36.03blitzragethere is only wireless here. I need to get some blank CDs, install Linux on here so I can develop and run Asterisk on it, and then I need a convertor to change the polycom cable to a cross over
15:36.11blitzrageso there are a lot of reasons why the phoen isn't up :)
15:36.44blitzrageI also need to find one of those little ziplink headsets so I can install a softphoen on the laptop
15:36.50blitzrageand I need more RAM too.... *grumble*
15:36.53ManxPower[TK]D-Fender, the biggest reason is we had to do it this way in the past and it works well for us.
15:36.58[TK]D-Fenderblitzrage : Don't carry RJ45 crimping gear with you at all times? :O
15:37.20ManxPowerAnd a box of Cat5!  Always have a box of Cat5 handy!
15:37.30blitzrage[TK]D-Fender: well I do have them, but I'm not recrimping my polycom cable :)
15:37.32*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
15:37.36[TK]D-FenderManxPower : Your "shared" phone implementation is reason enough :)  Only so many ways to operate that way with any sanity...
15:38.01ManxPower[TK]D-Fender, I hate it, but it fits the customer needs.
15:38.07blitzrageits one of the few things I hate about the polycom phone -- proprietary cable
15:38.14ManxPower[TK]D-Fender, also for most phones my macros handle it pretty well.
15:38.18[TK]D-FenderManxPower :  Get them seperate phones ... the cheap bastards!
15:38.34[TK]D-Fenderblitzrage : Proprietary my ass :)
15:38.54[TK]D-FenderSTANDARD full gigabit / PoE wiring :)
15:38.56*** join/#asterisk Seyr (n=kaladin@cpe-67-10-139-141.houston.res.rr.com)
15:38.59blitzrage[TK]D-Fender: well... close enough -- inline PoE on the cable and the one RJ45 has a "key" on it
15:39.17ManxPoweri.e.: http://pastebin.ca/33187
15:39.18blitzrageI can't go to a store and buy one of those cables
15:39.32SeyrI cant get phones to connect to * if the IP is a secondary IP. Anyone have any idea?
15:39.40[TK]D-Fenderblitzrage : Yes you can.  its a standard cable and have used ALL kinds and have wired my own...
15:39.58ManxPowerThe polycom 500s have a keyed jack on the phone.
15:40.03blitzrage[TK]D-Fender: sounds like a bitch -- and I can't find the inline power adapter anywhere
15:40.10ManxPowerThe 300s do not.  I don't know about the 600s, I doubt it.
15:40.17blitzrageI have an IP500
15:40.57[TK]D-Fenderblitzrage : You mean you want to MAKE a PoE custom cable for your own injection purposes?  thats different
15:41.10blitzrage[TK]D-Fender: no -- you said you do it all the time
15:41.15[TK]D-FenderI though you were talking about just using CAt5 for data and the power brick seperate
15:41.27ManxPowerThere is no reason to use a custom cable for the 500s
15:41.48ManxPoweruse the one that comes with it.  If you need a longer cable, longer cable into the existing polycom cable.
15:41.54*** join/#asterisk tengulre (n=tengulre@222.90.170.155)
15:42.03SeyrI'm trying to use Heartbeat with Asterisk and have the proxy IP set to come up as eth0:0 and if the IP is "eth0", it works just fine, but when I set it with heartbeat and it comes up as "eth0:0", the phones wont connect
15:42.05ManxPowerBTW, the place where you plug the power into the cable is CLOSEST to the wall.
15:42.09blitzrage[TK]D-Fender: uhhh... no -- how do you plug in the power brick into the polycom IP500? I don't see a power insert
15:42.24blitzrageManxPower: the problem is its not a crossover
15:42.36ManxPowerblitzrage, Um, make the extra cable be crossover.
15:42.44blitzrageManxPower: what extra cable?
15:43.11coppiceI have power over ethernet. when I pull that RJ45, the bugger stops dead in its tracks :-)
15:43.16ManxPowerblitzrage, all polycom 500s ship with a special polycom cable that has a plug on one end, a jack on the other end.
15:43.40blitzrageManxPower: I don't know what we're talking about the same problem
15:43.42ManxPowerif you want crossover just plug the crossover cable into the jack part.
15:44.03blitzragethe jack part?
15:44.17blitzragecable goes from computer eth port to phone -- its a straight through cable -- no worky
15:44.31blitzrageand I have to use the polycom cable since it has the inline power adapter
15:44.35ManxPowerblitzrage, you have the wrong cable then.
15:44.39blitzrageManxPower: I KNOW! :)
15:44.47ManxPowerblitzrage, why do you have the wrong cable?
15:44.48blitzragehence my problem
15:44.55ManxPowerThe IP 500 actually ships with TWO cables.
15:44.59blitzragebecause thats the cable that came with the phone
15:45.07[TK]D-Fenderblitzrage : you don't have the stanard wall-brick?
15:45.20blitzrageI go thte phone for free from someone and only have the straight through cable
15:45.33blitzrageI have the brick -- thats not the problem
15:45.50blitzrageI only have a straight through polycom cable -- and I need a xover
15:46.09blitzragehence my need for an adapter to convert it over -- and why I don't have the phone working on my laptop
15:46.16[TK]D-Fenderthen all you need is the brick and a straight cable for switches, and a cross-over standard lan for direct-to-NIC
15:46.31blitzrageI said that about 10 mins ago
15:46.52ManxPowerblitzrage, look at the cables on this: http://polycom.com/common/pw_cmp_updateDocKeywords/0,1687,699,00.pdf
15:47.04blitzrageI don't want a switch in the path
15:47.23blitzrageI just need the little adapter to plug onto the end of the cable to convert it
15:48.01[TK]D-Fenderblitzrage : I keep a short crossover patch and an RJ45 coupler for that reason
15:48.18blitzrage[TK]D-Fender: uhhhh yah -- hence why I don't have my phone working
15:48.27[TK]D-Fender:p
15:48.30blitzragethis has been a useless conversation :)
15:48.32[TK]D-Fender*sigh*
15:48.33ManxPowerblitzrage, is this the cable you have? RJ-45 -> bulge -> power bulge -> RJ-45
15:48.46ManxPoweri.e. the one that came with the phone
15:48.50blitzrageyes
15:49.20ManxPowerblitzrage, the non-power bulge, look CAREFULLY at it.
15:49.40blitzrageI only have one cable
15:49.47blitzrageand it has the power buldge
15:49.50blitzrageand its straight through
15:49.58ManxPowerso the cable only has 1 bulge (for power)?
15:50.06blitzrageyes
15:50.14blitzrageand the key RJ45
15:50.15file[laptop]over and over...
15:50.17ManxPowerAh.  all my IP 500 cable have 2 bulges.
15:50.31ManxPowerlooks like you have the real Polycom PoE cable, not the one that normally ships with the phone.
15:50.33file[laptop]so blitzrage, how's every little thing?
15:51.03blitzragefile[laptop]: eh
15:51.16ManxPowerblitzrage, on monday I can have someone ship you the correct cable if you pay the shipping.  We have extras.
15:51.47blitzragethats cool -- I'm just gonna buy a small adapter
15:52.19blitzrageshipping across the border would probably cost me liek $25 or something stupid
15:52.22blitzrageI hate customs
15:52.57file[laptop]blitzrage: you're in Florida, ManxPower is in the US...
15:53.40fugitivolucky blitzrage, poor manxpower
15:53.52blitzrageI can probably find a cable at Mix next week... if not... I'll get a hold of your ManxPower
15:53.59blitzrageyah... forgot I'm here for another week :)
15:54.05file[laptop]silly blitzrage
15:54.26blitzrageI get to avoid the snow for one more week
15:55.28file[laptop]oh, it'll get you
15:55.40file[laptop]just you wait...
15:55.47file[laptop]as soon as you get back, KABOOM - huge storm
15:56.01blitzrageyep == at least I work at home and don't need to leave the house
15:56.09blitzragemy office is on the other side of my bedroom
15:56.28SeyrI'm trying to use Heartbeat with Asterisk and have the proxy IP set to come up as eth0:0 and if the IP is "eth0", it works just fine, but when I set it with heartbeat and it comes up as "eth0:0", the phones wont connect
15:56.39Corydon76-homeHe says as the ice snaps his DSL line...
15:57.52blitzragedon't have DSL :)
15:58.00*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
15:58.05Corydon76-homeOkay, then, cable line...
15:58.12blitzragethat'd suck
15:58.34Corydon76-homeYep, trapped in a house without Internet access...
15:58.39file[laptop]"oh no! it found my weakness... it must be destroyed..."
15:58.56blitzrageoh well... I'd just go play drums or something
15:59.12file[laptop]1. ADSL 2. Cellular 3. Modem
15:59.13blitzrageok -- I'm off bor breakfast and to pick up my rental car
15:59.29file[laptop]blitzrage: don't crash... or stuff
15:59.58ManxPowerblitzrage, um, I'm moving next week.
16:00.21*** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com)
16:08.02*** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com)
16:08.49riddleboxis there a howto on setting up the voicemail access? I cannot find the default hunt group of the voicemail
16:09.16*** join/#asterisk dzlabing1 (n=dzlabing@h082218033038.host.wavenet.at)
16:09.34*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
16:09.39*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
16:11.59[TK]D-Fenderhunt group?
16:12.48*** join/#asterisk saftsack (n=oliver@p54A7F3E7.dip.t-dialin.net)
16:12.49saftsackhi
16:13.35*** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
16:13.46markithi, will svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 bring me the 1.2.1 as well?
16:14.32file[laptop]that gets you the latest 1.2 version with all the newest changes since 1.2.1 was released
16:14.58riddlebox[TK]D-Fender: sorry that is a term I am used to in programming other phone systems and voicemails
16:15.54[TK]D-Fenderriddlebox : What are you trying to do specifically?
16:16.16riddlebox[TK]D-Fender: I am trying to create an extension and then log into the voicemail for that extension
16:16.57markitfile[laptop]: and to have it automatically get everything in the 1.2.x?
16:17.13[TK]D-Fenderriddlebox : So you've hardly done anything in extensions.conf so far?
16:17.13riddlebox[TK]D-Fender: normally if we program a system, we create a group for voicemail ports to hunt in, then you call the pilot of the group to access voicemail
16:17.17markitso when 1.2.2 will be released, I will automatically get?
16:17.28file[laptop]markit: yes, if you update
16:17.41[TK]D-Fenderriddlebox : Forget old PBX terminology & methodology.  * is like striaght programming...
16:17.47saftsack[TK]D-Fender, hi i have a problem. if someone calls my second msn asterisk will answer the call and then the caller is able to call an interntal telephone. that works fine. but if the caller hangs up, before the call is answered the other telephone will ring until i kill asterisk
16:17.56markitsvn update instead of checkout, you mean?
16:18.05file[laptop]markit: make update
16:18.16riddlebox[TK]D-Fender: no I am reading through the AsteriskTFOT.pdf right now but it never mentions how to access voicemail, I am trying to create a sip extension as well
16:19.06[TK]D-Fenderriddlebox : Read up on the STDEXTEN macro.  that'll show you a way to dial an extension and have the caller be able to leave VM if busy or no-answer
16:19.25[TK]D-Fenderto pickup VM you'd have to make another extension.
16:20.58riddleboxso that other extension is what the user would dial to access the vm system then?
16:21.00file[laptop]food? food is a great idea!
16:21.01*** join/#asterisk toddf (n=toddf@ns0.fries.net)
16:21.47jimmy_deanPBSo I have the following 2 lines that applies for all incoming calls: "exten => s,2,GotoIfTime(8:00-17:00:mon-fri|*|*?regular-business-hours,s,1) exten => s,3,Goto(after-business-hours,s,1)" It always seems to do the GotoIfTime() but never the Goto() for after-business-hours...anything obvious I'm missing?
16:22.58*** join/#asterisk lilneon (n=tj_r3@pm4p2.cwdom.dm)
16:23.04lilneonhello everyone
16:24.24jimmy_deanPBAnyone have any ideas? Pretty quiet crowd today
16:25.00[TK]D-Fenderriddlebox : correct
16:25.34riddlebox[TK]D-Fender: one more question, is that the extension everyone on the system would use to access vm?
16:26.13lilneonyeah.... the usual talkative ones asleep or something>
16:26.14lilneon?
16:26.26file[laptop]riddlebox: let's do this... you have complete control over extensions, you can completely separate your asterisk into two companies if you wanted, and have different extensions for voicemail pickup
16:26.35[TK]D-Fenderriddlebox : Actually there are TONS of ways to make a call from anywherre to anywhere.  My STDEXTEN macro is heavily modded and I can have it so that if an EXT dials itself then it enters the VM box.  I also have it so that if you dial someones ext and hit * you enter the box as well.
16:26.55[TK]D-Fendertheir*
16:27.14*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
16:27.37riddleboxok I see, I have way more options to programming than I am used to
16:28.07file[laptop]yes and don't think of an extension as a phone, think of it as comprising of a list of instructions to execute
16:28.54[TK]D-FenderINFINITELY more.
16:30.39riddlebox[TK]D-Fender: if someone dials their ext and presses * does it ask for a password or does it just log them in?
16:30.48file[laptop]you control that
16:31.07file[laptop]it jumps to the a extension in the current context where you can do whatever
16:31.14riddleboxI got alot to learn/read lol
16:31.22ManxPower~docs
16:31.24jbotit has been said that 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
16:31.25file[laptop]yes
16:32.00[TK]D-Fenderriddlebox : Actually * will abort the Vm and jump to the "a" extension in the current context.  from there you can do whatever you want.
16:32.02saftsackhowto realize one internal dial string, if i have more than one telephone technics?
16:32.10[TK]D-FenderI have MINE go into the box to pick up messages.
16:33.00lilneonhey guys, has anyone here used satellite voip?
16:33.25[TK]D-Fendersaftsack : like "exten => 500,1,Dial(SIP/123&IAX2/456&ZAP/1,15) ?
16:33.43saftsackyes i thought that too, BUT
16:33.56saftsacki i want to call just one telephone
16:34.10saftsackthen i have a problem with this dialstring
16:34.28saftsackthe simplest solution would be to make dialstring per technic or?
16:36.41*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
16:36.48markitmmm http://svn.digium.com/svn/asterisk/branches/1.2.1 does not exist
16:37.00file[laptop]it's not a branch.
16:37.01file[laptop]it's a tag
16:37.12markitand how can I set a tag with svn?
16:37.18markitset=specify
16:37.26file[laptop]instead of branches, use tags
16:37.29[TK]D-Fendersaftsack : I have no idea what you are talking about.  Paste the dialsting ryou are having a problem with.
16:37.33file[laptop]http://svn.digium.com/svn/asterisk/tags/1.2.1
16:37.42markitah
16:37.47file[laptop]that will get you 1.2.1
16:37.51*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
16:37.52saftsackim searching for a dialstring
16:37.59saftsack<PROTECTED>
16:38.04saftsack<PROTECTED>
16:38.07saftsackhowto make 1?
16:38.34[TK]D-Fenderyou want to combine them?
16:38.39saftsackyes
16:38.39saftsackbut
16:38.46saftsackjust call a 1 on the beginning
16:39.00*** join/#asterisk javar (n=javar@69.79.133.185)
16:39.06saftsackok got it ^
16:39.24saftsackexten => _2.,1,Dial(misdn/4/${EXTEN}&(misdn/4/${EXTEN})
16:39.28saftsackis that right?
16:39.42meridalinuxcan someone tell me why I keep getting this error when I install zaptel
16:39.45meridalinux/lib/modules/2.6.13-15.7-default/build
16:39.45meridalinuxmake -C /lib/modules/2.6.13-15.7-default/build SUBDIRS=/usr/src/zaptel-1.2.1 modules
16:39.45meridalinuxmake[1]: Entering directory `/usr/src/linux-2.6.13-15.7-obj/i386/default'
16:39.45meridalinuxmake[1]: *** No rule to make target `modules'.
16:40.04sivanameridalinux: are you typing "make modules"?
16:40.16meridalinuxmake install
16:40.58meridalinuxI have recheck the prereq's
16:41.30saftsackexten => _1.,1,Dial(misdn/3/${EXTEN}&(misdn/4/${EXTEN})
16:41.33saftsacki meant this
16:41.47[TK]D-Fendersaftsack : more or less. but you should great a line group for those interfaces.
16:41.54saftsackyes ok
16:42.14saftsackISDNalle=misdn/3/*&misdn/4/*
16:42.19saftsackfor example such a variable?
16:42.21[TK]D-Fenderno, not like that.
16:42.28saftsack?
16:42.33markitany info about the availability of visdn for * 1.2?
16:42.48saftsack<PROTECTED>
16:42.49saftsackso?
16:42.51[TK]D-FenderIn your misn config file you should attribute # & 4 to a line group and like :  exten => _1.,1,Dial(misdn/g1/${EXTEN})
16:43.00saftsackok :)
16:43.44*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0it.dialup.mindspring.com)
16:44.06saftsackexten => _1.,1,Dial(misdn/g1/${EXTEN})
16:44.08saftsackok :)
16:44.29saftsackis this also possible with zapata.conf?
16:47.25*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
16:48.31[TK]D-Fenderthats how line groups work all around
16:48.39[TK]D-Fendergo read the wiki on whree to put that.
16:48.46riddlebox[TK]D-Fender: I think I see now, exten =>2223,2,VoicemailMain would the be correct?
16:49.34jimmy_deanPBanybody here good with the zaptel channels that can answer a question?
16:50.15mog_homejust ask jimmy
16:50.20jimmy_deanPBk
16:50.52*** join/#asterisk bweschke_ (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
16:52.16[TK]D-Fenderriddlebox : no, you would not go from dialing someon right into voicemail main.
16:52.20jimmy_deanPBI have a TDM400P card which I've used successfully in an old Asterisk box...so I copied the config from that * box into my new and faster box...incoming analog calls are seen and answered by asterisk just fine and my dial plan executes perfectly on the calls, but on the callers side it continually rings and you don't hear anything that * is playing from my dialplan almost like it never answered. This has be quite baf
16:52.20jimmy_deanPBfled. Any thoughts?
16:53.39mog_homeyou are using call progress no?
16:53.51jimmy_deanPBmog_home, you think I should be is what you're saying?
16:54.15saftsack[TK]D-Fender, signalling=fxo_ks
16:54.16saftsackgroup = 1
16:54.16saftsackchannel => 1 - 4
16:54.24saftsackthis is my entry in the zapata.conf
16:54.29jimmy_deanPBmog_home, I have it set to no currently
16:54.37saftsackbut if i call g1 just the first zap is called
16:54.56mog_homehmm thats odd
16:55.10mog_homeare you using r in your dial statements?
16:55.22jimmy_deanPBmog_home, yes I am
16:55.38saftsack[TK]D-Fender, do you have an idea?
16:55.47jimmy_deanPBmog_home, but I should hear a recording before any of the dial statements are executed which I'm not
16:55.52[TK]D-Fendersaftsack Those are phones?  not lines?
16:55.56jimmy_deanPBit doesn't even sound like asterisk picked up
16:56.13saftsack[TK]D-Fender, on each line there will be 1 phone
16:56.19saftsackso a linke will be a phone
16:56.27saftsackthe best would be if i can call
16:56.42saftsack5 and the second digit will be the line
16:56.49riddlebox[TK]D-Fender: I thought that was for someone to log into vm
16:56.56[TK]D-Fendersaftsack :  OH, ok forget everything we just did and use the Dial(phone1&phone2,15) style method we were just using
16:57.05saftsack;)
16:57.17saftsackhowto determine the second dialed digit?
16:57.22[TK]D-Fenderriddlebox : Have you read the STDEXTEN sample yet?
16:57.41saftsack${EXTEN} holds all digit, but howto determine the second?
16:57.49sylehmmm ${EXTEN:2:1} i would guess
16:58.03[TK]D-Fendersaftsack : just the second digit, or from the 2nd digit onwards?
16:58.05saftsackthanks i will test
16:58.15saftsackonwards?
16:58.17[TK]D-Fendersyle : I doubt he want jsut the 2nd
16:58.18riddlebox[TK]D-Fender: I am not sure where to find it
16:58.25saftsackyes
16:58.27saftsackjust the one
16:58.31[TK]D-Fenderriddlebox : its in the book, and on the WIKI.
16:58.58[TK]D-Fendersaftsack : then its ${EXTEN:1:1}
16:59.05*** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
16:59.05saftsackthanks :)
16:59.21[TK]D-Fendersyle : forgot its 0-based ;)
16:59.30syleis it?
16:59.32jimmy_deanPBmog_home, a weird problem, no? :)
17:00.10*** join/#asterisk Tili (i=Tili@202-133-67-221-dialup.sat.net.pk)
17:00.22[TK]D-Fendersyle : thats why exten => _9.,1,Dial(Zap/1/${EXTEN:1}) starts at the *2*dn digit :)
17:00.52syleyeah most arrays start at 0 doesn't surprise me
17:01.05mog_homei would think it might work with out the r
17:01.17*** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net)
17:01.21*** join/#asterisk Abbas (n=Abbas@203.128.19.51)
17:01.23mog_homebut its pretty odd you are having trouble getting the line answered
17:01.25saftsackworks :)
17:01.28saftsackthank you
17:01.31jimmy_deanPBmog_home, but the problem is before it even executes the Dial() command
17:01.33saftsackexten => _5.,1,Dial(Zap/${EXTEN:1:1}/${EXTEN})
17:01.36*** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com)
17:01.59*** part/#asterisk mhnoyes (n=mhnoyes@user-38lc0it.dialup.mindspring.com)
17:02.03Abbashello
17:02.05[TK]D-Fendersaftsack : No.. that line is very WRONG.
17:02.12sylelol
17:02.19saftsacki just have to delete the second exten?
17:02.28[TK]D-Fenderthe 5 is not supposed to be part of whats dialed.
17:02.31saftsackand substitute it with 1 or some other digit?
17:02.32Abbascan any one help   me ??    i want to use   chanunavail  in  extensions.conf
17:02.42meridalinuxI keep getting this error when I install zaptel 1.2.1 make install
17:02.43mog_homeid stick an answer in then
17:02.50saftsack[TK]D-Fender, yes thats right, but theres just one telephone on this line
17:03.02saftsackso it is not important what is called or?
17:03.29[TK]D-Fendersaftsack : exten => _5.,1,Dial(Zap/${EXTEN:1:1}/${EXTEN:2}) if you do 512345 it will dial 345 on line 2 <-
17:03.33meridalinux/lib/modules/2.6.13-15.7-default/build
17:03.34meridalinuxmeridalinux make -C /lib/modules/2.6.13-15.7-default/build SUBDIRS=/usr/src/zaptel-1.2.1 modules
17:03.34meridalinuxmeridalinux make[1]: Entering directory `/usr/src/linux-2.6.13-15.7-obj/i386/default'
17:03.34meridalinuxmeridalinux make[1]: *** No rule to make target `modules'.
17:03.47meridalinuxcan anyone help
17:03.48[TK]D-Fendermeridalinux : Stop spamming and use www.pastebin.ca !
17:03.52saftsackok thanks :)
17:04.17[TK]D-FenderBut your zap channels are PHONES, you shouldn't even be passing a # to them...
17:04.19meridalinuxbeen there.. don't want to spam just asking for help
17:04.41saftsack? ^^
17:05.41jimmy_deanPBmog_home, I do have an answer and I can see it getting executed from the console
17:07.06Abbascan any one help   me ??    i want to use   chanunavail  in  extensions.conf
17:07.42*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
17:09.25[TK]D-FenderDETAILS <-
17:10.02[TK]D-Fendersyle is full of snickers... but little advice :)
17:11.26jake1932Abbas: CHANNELUNAVAIL is in the DIALSTATUS variable - check that
17:12.01jake1932actually it's called CHANUNAVAIL
17:12.33ManxPowerAlso see README.variables in the Asterisk source and "show application dial" in the CLI and see extensions.conf.sample in the Asterisk source.
17:12.46[TK]D-Fenderjake1932 : He may be referring to ChanIsAvail command.. wait till he actually explains himself
17:12.49ManxPowerin extensions.conf.sample pay special attention to macro-stdexten
17:13.20jake1932don't know how i got 1932
17:13.48jake1932not quite that old
17:13.58*** join/#asterisk pengyong (n=lala@218.93.145.225)
17:14.09ManxPower"show applications" will list the available applications
17:14.44AbbasManxPower    thanks for suggession  ,   what i want to go is ,    i have  4 8 ports Gws  customer sending calls,   i want, if there is no channel on GW one call should go to second and same if no channel on second call should go on GW 3
17:15.00Abbas4, 8 ports GW
17:15.25[TK]D-FenderManxPower : Seen my STDEXTEN? :)
17:16.26Abbas[TK]D-Fender      no did not see this file
17:16.53*** part/#asterisk meridalinux (n=kmr@201.154.246.46)
17:17.31[TK]D-Fender[macro-stdexten]
17:17.31[TK]D-Fenderexten => s,1,Set(extension=${ARG1})
17:17.31[TK]D-Fenderexten => s,2,DBget(forward=EXT${extension}/FWDI)
17:17.31[TK]D-Fenderexten => s,3,GotoIf($[${forward}=NO]?DIAL|1)
17:17.31[TK]D-Fenderexten => s,4,Set(type=I)
17:17.32[TK]D-Fenderexten => s,5,Goto(fwd-${forward},1)
17:17.41*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
17:17.42[TK]D-FenderOMG
17:17.44[TK]D-Fendersorry...
17:17.53[TK]D-Fenderhttp://pastebin.ca/33194
17:17.56chapeaurougelol
17:17.59chapeaurougehi TK
17:18.02[TK]D-Fenderthe copy didn't take on my pastebin :)
17:18.36[TK]D-Fenderhey chapeaurouge
17:18.41Abbas[TK]D-Fender:  thanks    let me try to undersatnd it
17:18.49Abbasi am not good in it
17:19.04[TK]D-FenderAbbas : that wasn't for you, and if you need the kind of help I suspect it'd only confuse you MORE
17:19.47*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
17:20.19jake1932Abbas: you can dial each and then check DIALSTATUS for CHANUNAVAIL
17:20.53jake1932then move on to the next GW
17:21.08sylehow are you moving between GW's?
17:21.27jake1932trying a gateway using Dial? right?
17:22.03sylehope not
17:22.21sylethey can return busy , congestion etc to
17:22.28*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
17:22.35jake1932that's why you can check DIALSTATUS
17:22.49sylethats whats returning busy congestion etc hehe
17:23.00Abbasjake1932    just a min   i explain u in pastebin
17:23.46*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
17:23.53jake1932reading that is actually pretty funny - but i know what you meant
17:24.12*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
17:26.10ManxPowerAbbas, you should be able to use DIALSTATUS for that.
17:26.18jake1932lol
17:26.27syleprob make more sense to check active calls
17:27.57syleDIALSTATUS jumps around to much, i usually just isolate it in the h extension then loop it back to original context
17:29.07*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
17:29.28jake1932syle: shouldn't jump around - it's checking the status on the active call at the completion of a Dial statment.  unless i'm not understanding what you mean by jumping around
17:29.39jake1932s/call/channel
17:29.58syleit can jump to +101, or +2 or t, or i, or h etc etc
17:30.26jake1932syle: i think in 1.2.1 it always moves to the next priority by default
17:30.38syledoubt it
17:31.28syleeven if that is the case then that just means they removed +101 which still doesn;t help complex dialplans with timeouts etc
17:31.29Abbasjake1932:   http://pastebin.ca/33197
17:31.36Abbasjake1932:   http://pastebin.ca/33197
17:31.40Abbaspls check it
17:31.44*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
17:31.55bweschke_syle: not removed... you can always re-enable that behavior globally or per-application
17:32.17*** join/#asterisk meridalinux (n=kmr@201.154.246.46)
17:32.27jake1932Abbas: this is real simple
17:33.03Abbascan u pls make it  and paste in pastebin
17:33.26sylereason you loop it back to original context after h extension btw is otherwise you looose cdr record of dst of call if you don't
17:33.30jake1932Abbas: you have a bunch of Dial statements - after each one check the DIALSTATUS to see if it is CHANUNAVAIL
17:34.18Abbasjake1932:   i dont know the synteex and unable to make it after many tries after looking wiki help
17:34.36Abbasu please right few lines and paste at pastebin
17:35.42syleyeah jake can you!
17:35.44sylehahaa
17:36.00jake1932i'm not going to rewrite your dialplan for you
17:36.02*** join/#asterisk tobi__ (n=th@dslb-084-057-064-100.pools.arcor-ip.net)
17:36.25jake1932GotoIf($[$[DIALSTATUS}=CHANUNAVAIL]?true:false)
17:36.54jake1932false means it was busy or something else
17:37.09jake1932true means the chan was unavailable
17:37.14*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
17:37.18syleyou know
17:37.19jake1932both are priorities
17:37.24sylei wish there was a case statement
17:37.52jake1932nothing in ael like that?
17:38.19sylei don;t think you can realtime ael can you?
17:38.26jake1932don't know
17:38.38jake1932abbas: i'm sure you can figure this out
17:39.51syleif he knows basic programming sure, thats basically an if / else
17:41.00sylebut where are my damn else if's :)
17:41.27jake1932hmm
17:42.09*** join/#asterisk atif_ (n=atif@mbl-82-56-78.dsl.net.pk)
17:42.29Abbasjake1932:   i dont know much about programing
17:42.34jake1932it would be nice to not have to adapt everything
17:42.35Abbasthats y facing the problem
17:42.36syleright now all you can do is gotoif blah blah | gotoif blah blah
17:43.36syleit would also be nice to not have to write your own c module for everything
17:43.42syledo everything from dialplan
17:43.53atif_can some please tell me, how can I check ebitcount and crc4count values for a E1 span
17:44.05jake1932i haven't had to write a C module yet - luckily
17:44.32*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram)
17:44.32*** mode/#asterisk [+o kram] by ChanServ
17:44.40jake1932everything I needed was in there or pushed it out via AGI to another language
17:44.55coppiceatif: for a digium card just cat the /proc/zaptel/<whereever span> file
17:45.21atif_thanx coppice: I have tried that
17:45.26atif_let me check again
17:47.18atif_actually I want to check different values in zt_spaninfo
17:47.41atif_structure zt_spaninfo in zaptel.h
17:48.10atif_specially crc4count and ebitcount
17:48.33ManxPoweri.2 has a while statement
17:49.45jake1932hmm - looks like it has a case statment also
17:50.09jake1932according to http://www.voip-info.org/tiki-index.php?page=Asterisk+AEL
17:50.10atif_/proc/zaptel/span_no only tells span# is clear and inUse or not
17:50.20atif_no other information
17:52.41riddlebox[TK]D-Fender: in your macro-stdexten example can I copy that into my extenions.conf and  then build a [Main] and put my extensions in it or is there more to it?
17:53.30[TK]D-Fenderriddlebox : that paste I did was for ManxPower to look at.  Its considerably more than you shuold be considering at this moment and requires a lot of other scripts to setup the * DB
17:53.52riddleboxohh no I am on http://www.voip-info.org/wiki-Stdexten+macro
17:53.59coppiceatif: it also shows the error counts, if there are errors
17:54.16[TK]D-Fenderriddlebox : the only thing of use for you in there is the "a" part that dumps the caller into the VM box.  integrate that with the "reference" STDEXTEN.
17:54.38[TK]D-FenderOk, yeah read up on the WIKI's on and learn how to call it.
17:54.40*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167049176.nb.aliant.net)
17:55.09*** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com)
17:55.11riddlebox[TK]D-Fender:  hehe ok I was a little confused at first
17:55.26ManxPower[TK]D-Fender, I see your macro-stdexten and raise you my macro-stdexten!  http://pastebin.ca/33198
17:55.38*** join/#asterisk _victor (n=victor@193.226.149.70)
17:56.08jake1932ManxPower: was it easy to get the NV app?
17:56.19ManxPowerjake1932, yes.
17:56.44*** join/#asterisk bakermd (n=mbaker@exchange.i2telecom.com)
17:56.46jake1932have you tried any of the other ones like call progress detection?
17:56.50[TK]D-FenderManxPower : Nifty.... messy to read, but nifty :)  You should "abstract" it some more...
17:56.52ManxPowerIf you want spandsp, rxfax, txfax, and the two NV apps already patched in download http://www.fnords.org/~eric/asterisk.tar.gz
17:57.06[TK]D-FenderManxPower : Mine will likely become a bit more like yours before long.
17:57.14bakermdDoes anyone have VoiceMail working with RFC2833? Mine only works with Inband...
17:57.20jake1932that link be broken
17:57.32_victorhello all. i have been guided here by the h323 readme in asterisk, and i have a (hopefully) short question that i failed to get an answer for on google:
17:57.45ManxPowerAhrimanes,  I think it's http://www.fnords.org/~eric/asterisk.tar.bz2
17:57.57_victoris it possible to have a call from a sip channel to a h.323 channel in asterisk, in which the rtp will go directly between endpoints?
17:58.08jake1932yep - tnx
17:58.14ManxPowerbakermd, um, I have 70 phones using RFC2833, no VM issues.
17:58.20bakermdOkay
17:58.45_victori need this in order to call between a sip and a h.323 videophone and i think that this might be the only way, as asterisk will not understand and forward the video codecs required by the endpoints in the call setup
17:59.19ManxPower_victor, very few people do video with Asterisk.  Try searching the mailinglist archives or ask on the mailinglist.
17:59.36bakermdwell.. what I've got is an app that is answering, then prompting for input, followed by the #.  If I am Inband, I see the digits register in Asterisk as I type them.  In RFC2833, nothing appears, and I timeout after 20 sec.
17:59.55bakermdI figured VoiceMail handles this the same way I do
18:00.10ManxPower[TK]D-Fender, it's fairly abstract, notice my use of looping and "subscripts"
18:00.31ManxPowerbakermd, your phone is not sending RFC2833 digits then.
18:00.49ManxPowerthe PHONE and ASTERISK have to be set to the same DTMF mode or you'll experience things like you've seen
18:01.14bakermdI am calling from a cell phone, Level 3 passes the traffic to Asterisk
18:01.39ManxPowerbakermd, There's your problem.  I believe level 3 may only support inband.
18:01.43bakermda ha
18:01.52bakermdWell.. hmm
18:01.59bakermdCannot use G729 with inband
18:02.02atif_thanx coppice, thanx for ur help
18:02.04ManxPowerRegardless, if you change the dtmfmode= in asterisk, you have to get level3 to change it on your end too.
18:02.13ManxPoweron their end
18:02.14_victorManxPower, the only useful thing that i found is http://www.voip-info.org/wiki-Asterisk+video is it really that simple?
18:02.26[TK]D-FenderManxPower : I do.. but it just looks so unreadable :0
18:02.29coppiceatif_: can you see the errors now?
18:02.31ManxPower_victor, no idea.  Video is for marketing sissies.  Geeks use text mode.
18:02.35*** join/#asterisk sneak (n=sneak@datavibe.net)
18:02.39jake1932i think you can set dtmf=auto and it will take either
18:02.58ManxPower[TK]D-Fender, it will be much better when I switch it to AEL.
18:03.08ManxPowerjake1932, dtmfmode=auto must be new.
18:03.19bakermdI've never heard of it
18:03.34ManxPower[TK]D-Fender, I have accepted the fact that anything complex in extensions.conf is basically unreadable.
18:03.39[TK]D-FenderManxPower : and line 9 makes like 8 irrelevent :)
18:03.40jake1932This feature was added in CVS HEAD on Sep 6 2005 - yep
18:03.46_victorManxPower, video is very nice when you consider calling from voip to isdn.. it makes your customer to do many simultaneous isdn calls for only one video call :-)
18:04.01[TK]D-FenderManxPower : (virtually)
18:04.06ManxPower[TK]D-Fender, NVFax detect provides ringback?
18:04.10_victorthe issue is that video over ip to isdn gateways i found only for h.323.. and the nicest video phones are sip
18:04.15_victor:-)
18:04.21[TK]D-FenderManxPower : SORRY... IN TH _xxxx
18:04.36ManxPower[TK]D-Fender, Ah!  PRIORITY 9
18:05.06[TK]D-FenderThe fail-out option makes excessive gotos pointless.  htats why the typical STDEXTEN jump on the pririty and assumes non-busy = unavail
18:05.46ManxPower[TK]D-Fender, you'll notice the Gotos go to different priorityies.
18:05.53*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
18:05.55ManxPowerAnd if some day there's a new setting for DIALSTATUS......
18:06.27ManxPowerlike DIALSTATUS=ANSWER, for example.
18:06.31[TK]D-Fenderyeah, but in effect you account for just about every possibility except you only really filter on 1 (busy)
18:07.36*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
18:07.38[TK]D-Fendereek
18:07.42bakermdQuestion then: My app answers a call from level 3, then lets me provide a number to call out to.  I need the call out to be G729, so how can I get this to work... add lines to extensions.conf so that after the number is given to asterisk, force G729 and rfc2833, then place the call?
18:07.54filewasn't me
18:08.06ManxPowerbakermd, your extensive search of the mailing list about level3 was not helpful?
18:08.16MikeJ[Laptop]file, was too!
18:08.18[TK]D-Fendermy internet is shaky right now it seems
18:08.28bakermdOur account rep claims that they support it.. but I do not trust him now
18:08.37bakermdF'n sales people
18:08.44file[TK]D-Fender: it's you!
18:08.53file69873?
18:08.54ManxPower[TK]D-Fender, you'll notice that if DIALSTATUS is "${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION" then I loop to the next DIAL_DEST[INDEX]
18:09.03MikeJ[Laptop]hey hey file...
18:09.09MikeJ[Laptop]guess what...
18:09.12tzangerManxPower: what are you describing now?
18:09.15MikeJ[Laptop]:(
18:09.22ManxPowertzanger, my macro-stdexten
18:09.25tzangerahh
18:09.32tzangerthat is quite the macro I have to admit
18:09.33ManxPowerhttp://pastebin.ca/33198
18:09.41MikeJ[Laptop]the stuff I was talking about last night no longer segfaults....
18:09.41ManxPowertzanger, this one is much simplier.
18:09.48MikeJ[Laptop]now it just doesn't work!
18:09.50[TK]D-FenderManxPower : the only one you don't seem to imply is "ANSWER" which seems to hangup in my scenario anyways...
18:10.00tzangerno no I want macro-s00per-d00per-ultra-exten
18:10.06ManxPower[TK]D-Fender, ANSWER falls thru to the Hangup
18:10.27[TK]D-FenderMine seems to do that without explicitly doing that already ....
18:10.33[TK]D-Fendernot sure why now that I think about it...
18:10.54[TK]D-Fenderbut its what the sample one does too...
18:10.55ManxPower[TK]D-Fender, maybe when a call is answered your macro falls off the end?
18:10.56file[TK]D-Fender: phone tag, you're it
18:11.14ManxPowerI could have accomplished the same thing with a priority gap
18:11.18[TK]D-Fenderfile : RING!
18:12.39ManxPowerBut the Hangup makes it explicit
18:12.43ManxPowerrather than implied.
18:13.42ManxPowerResults 1 - 10 of about 69 from lists.digium.com for  level3 dtmf. (0.33 seconds)
18:16.19*** join/#asterisk cnet2 (n=jjohn@201.192.107.58)
18:16.43cnet2does all asterisk have a gui interface? or just the xorcomm? what-s the command to access it?
18:16.50foo8armoh crashes my *. im not using it specifically starting * spawns a mpg123 process  (YES i have a zaptel device for timing)
18:17.44Qwellcnet2: You have some reading to do...
18:17.46jake1932foo8ar: what version?
18:17.53jake1932of asterisk
18:17.55cnet2jeje ok
18:17.57foo8ar1.2.0
18:18.06jake1932you can use format_mp3
18:18.10ManxPowerfoo8ar, and you have mpg123 v0.59r?
18:18.15jake1932it's in asterisk add-ins
18:18.36jake1932just load format_mp3 in modules.conf before you load musiconhold
18:18.52bakermdApologies ManxPower - did not do my homework
18:19.05foo8armpg123: 0.59s-mh4
18:19.05jake1932and use native (at the bottom of the example musiconhold.conf)
18:19.17ManxPowerfoo8ar, there's your problem
18:19.24foo8arok
18:19.28ManxPowerEither do what jake1932 said or use the correct version of mpg123
18:19.28*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
18:19.56ManxPowerin fact, you should uninstall your current mpg123, then in the asterisk source do a "make mpg123" that will download and build the correct version.
18:19.57foo8arwhat does format_mp3 do?
18:19.57*** join/#asterisk Assid (n=assid@203.115.64.62)
18:20.04Qwellplays mp3s?
18:20.11Assidheya
18:20.14Assidsup Qwell
18:20.42jake1932you can use format_mp3  instead of mpg123
18:21.21Assidumm.. if i use iax2 show netstats .. and i fi see lost and droppped on the local side
18:21.23Assidwhat does it mean?
18:24.44*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
18:27.04Assidanyone?
18:29.17riddlebox[TK]D-Fender: do you mess with broadvoice or vonage at all?
18:29.45Qwellvonage won't work with *, broadvoice rarely does
18:30.03[TK]D-Fenderriddlebox : nope.  I do POTS @ home, PRI at work, but trunk SIP from work -> home
18:31.06jake1932i had broadvoice work with the BYOD plan - but the quality wasn't consistant - not sure if that's what you were referring to
18:31.42Qwelljake1932: They seem to have gotten a little better, but 3-4 months ago, somebody would come in and complain about broadvoice daily
18:31.50file[TK]D-Fender cheats!
18:31.52riddleboxQwell: there are tech shows like the linux link tech show that use broadvoice, my question though is, can you answer a call with asterisk then forward it to broadvoice's vm?
18:35.47jake1932riddlebox: is the call coming from broadvoice?
18:37.54riddleboxyes, I want to call in and have an asterisk voicemail menu answer to give me a couple of options then if I want to leave a message go to the broadvoice voicemail if I want to leave a message
18:38.13Qwellwhy use the bv voicemail?
18:38.56foo8arManx: that did it. THX
18:39.10riddleboxI will have to access the asterisk vm by sip phone, I will not be able to access it from my phones
18:39.23Qwellriddlebox: set it up so you can
18:39.32Qwellit's not difficult
18:39.58riddleboxQwell: how I do not have any digital phones, just an analog phone that is connected to the broadvoice box
18:40.47jake1932broadvoice box?  an ATA?
18:41.07riddleboxit is whatever they sent me to plug a phone into
18:41.13jake1932an ATA
18:41.31jake1932setup your ATA to hook up to asterisk and use asterisk to direct calls out to bv
18:41.52riddleboxhow do I setup the ATA to work with asterisk?
18:42.07Qwellugh
18:42.08Qwellhack
18:42.17*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
18:42.25AssidQwell: could you help me with that iax2 show netstats
18:42.26jake1932what ATA?
18:42.31Assidwhats the remote / local side mean
18:42.34riddleboxit is a Supra
18:42.40jake1932Supura what?
18:42.49QwellAssid: dunno
18:43.06riddleboxit is sipura 2100
18:43.52jake1932<PROTECTED>
18:44.30jake1932i think there's even a doc on the sipura site on how to do it
18:44.47Assidhrmm how do you hang up someone whose in the IVR menu on incoming call through IAX?
18:44.59ManxPowerAssid, the Hamgup application
18:45.12Assidhangup doesnt work :(
18:45.35AssidNo such command 'hangup' (type 'help' for help)
18:46.02Qwelldialplan
18:46.20Assidhes already inside.. i wanna hang him up
18:47.31file[laptop]soft hangup
18:47.34file[laptop]if you wanna do it from the CLI
18:48.01riddleboxsweet I will look it up
18:48.18Assidthanks file
18:51.10jake1932riddlebox: if you're still around - you can try this link: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+SPA-2000
18:51.21jake1932should be similar
18:51.32QwellYou're assuming it isn't locked
18:52.02jake1932true - if it's locked - well.. ummm
18:52.51jake1932unlock it?
18:53.12jake1932or get a new ATA
18:53.23Qwellor, just get a noncrap provider
18:53.54jake1932even if he does that, still stuck with a potentially locked ATA
18:54.48*** join/#asterisk coppice (n=chatzill@142.198.17.210.dyn.pacific.net.hk)
18:58.21atif_coppice: I dont see any errors, no crc no ebit ..... but are you sure, I will see them in /proc/zaptel/* if there are any errors
18:58.45coppiceunless someone has removed them recently.
18:59.49atif_coppice: how one can remove, these are just counters
18:59.52*** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca)
18:59.55atif_or they can be reset
19:02.24*** join/#asterisk f0urtyfive (n=noone@71.225.226.175)
19:03.49coppiceit looks like only the E400/T400 cards implement that. DUMB
19:04.46riddleboxjake1932:where can I get an ATA that will work with asterisk?
19:04.55coppicestrange. I thought the other cards used to implement the error counts
19:05.22Qwell:(
19:05.55blitzrageI have a car now
19:06.01Qwellooo
19:06.04blitzragenot that there is anything to do around here
19:06.07blitzrageMiami is boring
19:06.20Qwelldrive to CA :p
19:06.27blitzragehahahaha
19:06.31blitzragecan't take the rental out of the state
19:06.49Qwellsays who? ;]
19:06.49bakermdMiami - the place where parties last from 7:00 PM to 11:00 AM - there is nothing to do??
19:06.58coppicei thought rentals were always registered in another state
19:07.06jake1932riddlebox: i haven't found one i liked.  I just got a VOIP phone
19:07.06Qwellcoppice: eh?
19:07.33riddleboxahh ok
19:07.35jake1932maybe someone else can recommend one that worked for them
19:07.52QwellDid you try using your spa2100?
19:08.24coppiceI don't think i've rented a car in the US that was registered in the state where I rented it. I understand there is some tax dodge related to that
19:08.59blitzragecoppice: who knows -- either way, they made it very clear that car can't leave the state
19:09.03QwellI don't think I've ever seen a CA rental car that wasn't registered in CA
19:09.14atif_zap show status,,,gives crc4 count
19:09.29atif_but I am unable to find ebit count till now
19:09.50Qwellblitzrage: Did they say it can't leave the state, or you can't drive it out?
19:10.01Qwellyou could always tow it, if it were the latter
19:10.04coppiceatif: where does zap give the CRC4 count? what card are you using?
19:10.29atif_its quad port TE
19:11.07blitzrageQwell: the car can't leave the state
19:11.17Qwellholy crap, VON is expensive
19:11.19blitzrageQwell: although with what states border FL -- who the hell woudl want to? :)
19:11.37blitzrageQwell: uhh... yah... you're caught up to the year 2002 now...
19:11.41ManxPowerQwell, I don't think VON is worth the money.
19:11.47blitzrageManxPower: it certainly is not
19:11.49ManxPowerIt seems to be a Show for Suits.
19:12.04blitzrageVOn is boring -- if you're a techy -- don't go
19:12.21Qwellwhat other good conferences are there?
19:12.30ManxPowerAtricon!
19:12.35Qwellbesides that
19:12.44blitzrageInternet Telephony?
19:13.11*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
19:13.15blitzragenever been -- but going in Jan.
19:13.50QwellWhere's that?
19:14.40file[laptop]there is nothing... absolutely nothing... on
19:15.18jake1932when is there going to be something near the Phila/NY area?
19:15.28Qwelljake1932: never
19:15.34coppiceatif_: looking in the source code for zaptel, the only card which supports the error counts is the T400/E400. Where do you see CRC4 error counts displayed?
19:15.45jake1932why?
19:15.49*** join/#asterisk netwind (n=netwind@gprs.rekom.ru)
19:20.34blitzrageanyone have a motivation pill?
19:21.03atif_when I do "zap show status" on asterisk console it gives crc4 count ..... but its 0
19:21.21Poincareis it possible to have a 'iax2 debug' for one account/peer only?
19:21.35coppiceits lying. the only card implementing these counts with now is the t400/e400
19:21.40atif_let me show u in the private window
19:21.50QwellPoincare: sure
19:22.01PoincareQwell: do you know how?
19:22.04coppicethey will show up in the /proc/zaptel/xxx files for one of those cards
19:22.04Qwellerr, I thought so
19:22.12blitzrageQwell: nope -- not in the iax channel
19:22.25blitzrageQwell: I had file[laptop] make a patch to make it work for me ... but we've both lost the patch :)
19:22.29Qwellheh
19:22.46blitzragethat iax channel keeps falling futher and further behind SIP
19:22.49file[laptop]we're silly and don't believe in backups
19:23.06blitzragecool -- back to the future is on
19:24.24*** join/#asterisk Vendetta9 (i=Vendetta@209-112-190-184-cdsl-rb1.anc.acsalaska.net)
19:24.34file[laptop]is that REALLY cool? or simply, imaginary cool
19:24.39Poincarewith 5 iax2 peers, debug is impossible to read
19:24.46blitzragepainyeppers
19:24.54blitzragedamn latency -- makes typing hard :)
19:25.01blitzragePoincare: yeppers
19:26.03blitzrageuse ethereal
19:26.05*** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
19:26.11Ferrarigood day all
19:27.57jake1932Ferrari: will redirect work?
19:28.31Ferrariwell redirect can grab a leg from the meetme but then it would just ring my phone
19:28.48*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
19:28.57Ferrariand i want to be able to have the call to be a live bridge
19:29.03Ferrariuser 1 picks up phone
19:29.11Ferraridials agi extension
19:29.34Ferrariand wolla agi grabs a leg from the meetme and bridges it to the caller
19:29.41Ferrarino ringing of phones etc
19:29.50Ferrarii saw something about changrab
19:30.05Ferraribut can't find it, and not sure it will work in version 1.0.10
19:30.58jake1932why would the caller just be put in meetme - maybe i''m missing something here
19:31.06jake1932s/would/wouldn't
19:31.13Ferrariwell i need to pass dtmf through to the other end
19:31.20Ferrariand meetme clamps it
19:33.24file[laptop]theoretically if your DTMF was inband, you might have a chance
19:33.24jake1932your original q was to "grab a call out of a meetme room and bridge to it" bridge to the caller you just grabbed from the meetme room?
19:33.42Ferrariyes
19:33.52Ferrarii am extensoin 100
19:34.00Ferrarii want to bridge to sip/200
19:34.04Ferrariwho is in a meetme
19:34.17Ferrari100 dials *999<200>
19:34.28file[laptop]easier said then done
19:34.29Ferrariand wolla Sip/100 and Sip/200 are chatiing
19:34.32Ferrariin a simple since
19:34.37Ferrariagreed
19:35.02jake1932is 100 and 200 still in the meetme at that point?
19:35.22jake1932or are they out of the room?
19:35.33Ferrari100 was never in the room
19:35.42Ferraribut wants to link to 200
19:35.46Ferrariwho is in the room
19:35.58jake1932ok
19:36.13Ferrarii saw some manager bridge work in mantis, but looks like it is now all for 1.2
19:36.47Ferrariand looked like someone had something called app_changrab thqat did what i need but that is no where to be found
19:37.06jake1932hmm
19:37.12Ferrariso i am hoping someone in here has all the answers *hint *hint
19:37.32Ferrarifile[laptop] any ideas
19:37.34Ferrari??
19:37.56Ferrarii always seam to find the weirdest shit to want to do
19:38.12jake1932i know this is a little wack, but could you redirect out 200 to a park and have 100 redirect to pick up the park?
19:38.48file[laptop]I'm trying to think of how you could do it in the core, it may be possible
19:39.10Ferrariyeah but that would mean i need to know what parking space they were in and a lill hackier than i want
19:39.20Ferrarithat would rock file
19:39.21file[laptop]it's already going to be a hack
19:39.27jake1932yeah - no doubt
19:39.28Ferrariagree
19:39.44Ferrarijust like saying the word hackier
19:39.48Ferrarisorta rings
19:39.50Ferrarihuh
19:40.12file[laptop]go for app_changrab
19:40.22file[laptop]you might be able to change it... to the exact behavior you want
19:40.28file[laptop]as it does it via CLI right now
19:40.29Ferrariif i could find it
19:40.43Ferrariapp_changrab all links are dead
19:40.46file[laptop]ah lemme see here
19:40.55Ferrariunless you can post a working one
19:40.58manygood that i dont understand the task at all
19:41.31Ferraritask=something better left until tomorrow when you can play today
19:41.34Ferrari;)
19:41.41SkramXI want a Ferrari
19:41.48Ferrariso do i
19:41.52manyiam just playing, so? ;)
19:41.52SkramX:)
19:41.53Ferrariatleast i have the nick
19:42.25Ferrar1I have a "leet" Ferrari
19:42.31file[laptop]I'm grabbing a copy for you Ferrari
19:42.41Ferrariyou are the best
19:43.24file[laptop]http://quark.file-radio.com/asterisk/app_changrab.c
19:44.03Ferrarithanks
19:44.09Ferrarinow for the compile
19:44.15Ferraribrb
19:44.42*** join/#asterisk areski (n=areski@88.5.208.72)
19:44.52areskihello all
19:45.01SkramXHiya
19:45.40areskiquick kick question : someone knows if we can find asterisk files encoded in g723
19:46.27shido6yes
19:47.22areskioh shido
19:47.39areskiso tell me :)
19:48.30Vendetta9Is anyone familier with using a D-link DVG-1120s as a client with *?
19:48.48Ferrariok well i uncommented the AST_10_COMPAT and tried to complie
19:48.49moralewhats a DVG-1120?
19:48.51Ferrariand no dice
19:48.57*** join/#asterisk kernoman (n=bryan@82-33-9-8.cable.ubr14.newt.blueyonder.co.uk)
19:49.46kernomanhow do i set the maximum length a call can last for, i want to be able to automatically disconnect my pstn line if its in use for more than an hour
19:49.58jake1932Ferrari: if you did want to go the park/unpark, I think ValetParkCall and ValetUnParkCall will let you specify and determine the "parking spots"
19:50.23Qwellkernoman: L option to dial
19:50.45kernomanQwell: can you clarify that a bit please?
19:50.48Ferrarihttp://pastebin.com/458707
19:50.52Qwellshow application dial
19:51.33kernomanwhere do i put that in my config so its a default setting?
19:52.28Vendetta9morale:  a DVG-1120 is a D-link VoIP gateway.  comes setup with ATT calladvantage settings.  Default uses MGCP but can be unlocked to use SIP.
19:54.44kernomandumb question - ignore it :)
19:55.18Qwelldid
19:57.09moraleVendetta9: ah thanks
19:57.19moraledid you unlock it to use SIP?
19:57.28moraleif it can use sip then you can interface it with asterisk
19:57.43Ferraribrb
19:58.10kernomanjust to clarify that the syntax would be Dial(L(x[:y][:z] rest of dial string )
20:00.28kernomanor is it appended to the end of the Dial command?
20:01.05file[laptop]end
20:01.09ManxPowerkernoman, "show application dial
20:01.10Vendetta9anyone use a DVG-1120 as a client?
20:01.32jake1932Vendetta9: have you tried it yet?
20:02.00kernomanthanks
20:02.15kernomanhappy christmas
20:02.51*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
20:03.44kernomandoes ms mean miliseconds?
20:04.15ManxPowerkernoman, usually
20:04.35kernomanManxPower: Cheers
20:04.59Ferrarifile do yuo know if that version of changrab is compatible with 1.0X
20:06.00file[laptop]I don't touch 1.0 stuff
20:06.08Ferrariotcha
20:06.13Ferrarigotcha even
20:07.09Vendetta9jake:  yes I tried it.  Problem that I'm having is that no matter what settings I put in it will not register with asterisk.  tells me username/auth mismatch.  I'm using the same user info from a softphone that I have running on this machine (with softphone logged out of course)
20:07.34jake1932are you running it in a local network?
20:08.02jake1932or to a remote asterisk server over the internet?
20:08.39Vendetta9local
20:08.57jake1932can you take out the password requirement for the phone on asterisk?
20:09.14Ferrariwell shit fire it will not compile
20:10.15Ferraripisses and moans about _bridge
20:10.25Ferrarithanks for the effort anyway
20:14.11jake1932Vendetta9: use something real generic: http://pastebin.ca/33224
20:14.19jake1932for ext 200
20:14.49Vendetta9jake doesn't seem to make a dif
20:15.51jake1932Vendetta9: if you're getting "username/auth mismatch", it means it's failing to authenticate
20:16.13jake1932by adding a generic entry in sip.conf, you can make it real easy
20:16.18Assidhrmm.. im trying to override cdr.. but it doesnt seem to work
20:16.33Assiddamn.. its read only
20:16.52jake1932Vendetta9: you can also do a sip debug from the CLI to get a better idea of what's going on
20:17.01*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
20:17.18file[laptop]why does the Discovery Channel have The Amazing Race....
20:17.43jake1932because every network needs at least 1 reality show
20:17.49[hC]Err... Is there some known issue with not being able to dial Voicemail(u${EXTEN}) in 1.2.1 ?
20:17.50file[laptop]evil
20:17.54*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
20:17.59[hC]it shows it execute in the dial plan, but seemingly just skips it over.
20:18.18jake1932yes hC
20:18.24jake1932it's different syntax
20:18.39[hC]Ah, of course. I should have known. :)
20:18.53jake1932exten,u now
20:18.59[hC]its Voicemail(box,u)
20:19.00[hC]yeah
20:19.05[hC]Thanks... :|
20:19.08jake1932np
20:19.15file[laptop]pure sillyness
20:19.38[hC]a console debug message would have helped :)
20:19.40file[laptop]and is it evil that I'm rooting for all of these opponents to run into critical problems in this show?
20:19.55file[laptop]I'm actually hoping nobody gets on this flight just to see them freak out
20:21.30*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net)
20:22.14moraleniZon: do you work for shaw?
20:22.50[hC]jake1932: do i need to change voicemail.conf or something as well? just changing it to Voicemail(ext,u) doesnt seem to have helped
20:23.01Qwellold style parsing still works
20:23.19[hC]<PROTECTED>
20:23.21[hC]skips right over that
20:24.57Qwell[hC]: I don't know if it's the same, but do you use realtime voicemail users?
20:25.06[hC]Qwell: nope, no realtime here..
20:25.10Qwellk
20:25.31Ferrarifile was there anytype of header file for changrab
20:25.35[hC]voicemail.conf must have changed.
20:25.36[hC]Dec 10 12:24:29 WARNING[23624] app_voicemail.c: No entry in voicemail config file for '103'
20:25.41Qwellheh
20:25.49Vendetta9jake  thanks but still no dice.  I used that generic entry but am still getting the same error.  the error is at http://pastebin.ca/33226
20:25.55[hC]unless i HAVE to specify context now
20:26.00Ferrarithere is a switch define but nowhere is there a definition on what to do if defined
20:26.06Qwell[hC]: You do...if it isn't in default
20:26.14QwellOR
20:26.25Qwellgeneral, searchcontexts=yes
20:26.56Qwellsame cause, different issue :p
20:27.18[hC]sweet, i added that, now when idial in, * crashes.
20:27.21[hC]haha
20:27.40Qwellsame cause, same issue :p
20:28.09Qwellyeah, it's kinda funky
20:28.14[hC]if i use the context in Voicemail() it works. using searchcontexts does not.
20:28.17Qwellthat patch broke realtime voicemail users...
20:28.21[hC](in that it makes asterisk die)
20:28.22EriSanhey guys, when i try to call i see this in the CLI: == Everyone is busy/congested at this time (1:0/0/1)
20:28.39[hC]EriSan: It means whoever you are dialing is marked as unreachable
20:28.39EriSanany idea? (lines are not busy)
20:28.40Qwell[hC]: mind a msg?
20:28.43[hC]either in sip, or iax.
20:28.46[hC]Qwell: shoot
20:31.21*** join/#asterisk Tili (i=Tili@202-133-67-75-dialup.sat.net.pk)
20:31.53Tiliwhen Asterisk sends a SIP INVITE. does it open the RTP port at that time or it waits till ACK is received from other peer
20:32.17Tilimy Eusso ATA tells me RTP Port not opened and so it sends back with 606 Not Accepted.
20:33.52*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
20:36.03*** join/#asterisk dardavoip (i=dardavoi@84.22.47.106)
20:36.28*** join/#asterisk Renacor (i=vircuser@ppp-71-133-4-237.dsl.irvnca.pacbell.net)
20:37.47Renacoris there a way to turn off the queue position status and tell people an approximate wait time instead? like instead of "You are number 5 in the queue" to "Your call will be answered in approximately 8 minutes" ?
20:37.48Qwellsome people ignore /me
20:37.50Qwelljust type
20:38.12file[laptop]dardavoip: this isn't the place for that
20:40.03*** join/#asterisk lofi-rev (n=lofi@198.145.218.209)
20:40.23*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
20:40.28lofi-revWhat command do I use to ring multiple extensions at once?
20:40.34QwellDial
20:40.57QwellDial(blah/1&blah/2)
20:41.30dardavoipanyone now where can i buy premium routes ?
20:41.38Qwell<file[laptop]> dardavoip: this isn't the place for that
20:41.41lofi-revQwell, thanks
20:41.58file[laptop]and repeating it just makes me sad
20:41.58dardavoipQwell where can i find please for that ?
20:42.10Qwellgoogle?
20:42.11file[laptop]try the asterisk-biz mailing list
20:42.18file[laptop]or like Qwell said, just Google it
20:42.24dardavoipok thenks
20:43.34wshsinbound calls via sip from broadvoice are getting busy tone, and nothing's shown in the logs, even with verbosity set at 10. when i use sip hardware in place of asterisk for broadvoice, inbound calls work. what would cause such a problem?
20:43.55wshs(and it's only a problem with broadvoice, no other providers)
20:44.16file[laptop]wshs: you might not be registered, might not be properly setup in sip.conf, might not be going to the right context
20:44.54wshswould 'sip show peers' show if im correctly registered?
20:45.00file[laptop]sip show registry
20:45.05wshsah, k
20:45.27*** join/#asterisk backblue (n=moo@87-196-6-83.net.novis.pt)
20:46.31lofi-revTwo questions, the answer to the first might answer the second, where can I find a listing of all the commands available for use in extensions.conf? and how do I specify the number of times to ring a device before moving on to the next command?
20:46.40[hC]cute.. my sip show registry shows nothing ;)
20:46.46[hC]yet i have like... 20 devices
20:47.02wshsfile[laptop], that says im registered.
20:47.59wshsif there was a 'disconnect' between the register and the [..] in sip.conf, wouldn't it complain in the console?
20:48.00file[laptop]okay then do a sip debug
20:48.03wshsk
20:48.06[hC]nvm.
20:48.06file[laptop]call your number
20:48.10file[laptop]and see if anything comes out on it
20:48.27[TK]D-Fenderlofi-rev : www.voip-info.org has the full list of commands.  For the dial time its "exten => 1234,1,Dial(SIP/100,[howlongtodial])
20:48.44*** join/#asterisk marc32422 (n=marc3234@206-248-157-52.dsl.teksavvy.com)
20:49.00lofi-rev[TK]D-Fender, thanks
20:50.52*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
20:50.56file[laptop]Matttttt
20:51.06mog_homefileeeee
20:52.37file[laptop]mog_home: are you up to no good?
20:52.42Qwellwhen is he not?
20:52.50file[laptop]when he's asleep maybe?
20:53.01Qwellthen he's dreaming about no good
20:53.15file[laptop]dreaming doesn't count
20:53.24Qwellok then
20:53.47bsdfreakheh
20:57.53Renacoris there a way to turn off the queue position status and tell people an approximate wait time instead? like instead of "You are number 5 in the queue" to "Your call will be answered in approximately 8 minutes" ?
20:58.20Qwellthere are two separate config options for those
20:58.28Qwellannounce-position and announce-holdtime
20:59.22mog_homehmmm i dont know of this good you speak of
21:00.20Renacorthink i figured it out thanks
21:01.35mog_homeand how can i get up to it
21:01.52Qwellmog_home: it costs money
21:02.02QwellI can get you up to it for a small fee
21:02.23mog_homehmm sounds pretty nice...
21:03.46file[laptop]mog_home: I also have some plutonium for sale
21:04.48Renacorcareful the fbi is listening
21:05.14file[laptop]meh I have clearance
21:05.19file[laptop]O.O
21:05.24Qwellcanada doesn't extradite
21:06.01*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
21:06.17mog_homewhat canada are you talking about
21:06.28mog_homethey extridite always
21:06.39mog_homemost recently that soldier
21:06.48Qwellmeh :p
21:07.32*** join/#asterisk mrgoby (n=mrgoby@pcp05307400pcs.wanarb01.mi.comcast.net)
21:07.50mrgobydoes anyone know what "call failed to go through, reason 0 " mans ?
21:07.51*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
21:08.05mrgobymans == means
21:08.05*** part/#asterisk bweschke_ (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
21:08.14Qwellmeans the call couldn't go through
21:08.43mrgobyok, does anyone know what reason 0 is ?
21:09.04mrgobyor, is there a lookup table of the reasons somewhere ?
21:10.38mrgobyit seems also that the iax module is complaining about "iseq 5 not in window"
21:10.46wshsfile[laptop], maybe i'll get to that debugging once i can find my cell phone :(
21:11.39*** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl)
21:13.34*** join/#asterisk _victor (n=victor@193.226.149.70)
21:13.37*** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
21:13.54*** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
21:14.31_victorhello.. i am trying my first steps into asterisk, and i am stuck in a simple thing: trying to register sjphone with it. is it appropiate to paste more details (logs, configs) here?
21:16.50mrgobyvictor: pastebin.ca is a good place for config posting
21:17.02mrgobyjust dont put your passwords in there, obviously :0)
21:17.09_victorbasically i am gettying Username/auth name mismatch when trying to register
21:17.48mrgobypaste your sip.conf
21:18.11*** join/#asterisk Abbas (i=Abbas@202.147.162.98)
21:19.04Abbashello twisted
21:19.07mrgobysoooo...  why is pbx_spool barfing  ... reason 0 ?
21:19.17Duttshi guys what is the svn command to get the release 1.2 code?
21:19.23Dutts1.2.1 I mean
21:19.34mrgobysvn co URL
21:19.58Duttswhat is the url of the release? asterisk website gives one to get latest snapshot of release branch, but I want the actual release...
21:20.25mrgobyif you want cutting edge, i would imagine that is trunk
21:20.55Qwellhttp://svn.digium.com/svn/asterisk/tags/1.2.1/
21:21.14DuttsQwell: is this the 1.2.1 stable release?
21:21.16Qwellyes
21:21.19Duttscheers =)
21:21.20Qwelltags are releases
21:21.29Qwellthe same goes for zaptel, libpri, etc
21:21.36Duttsthey all up to 1.2.1 ?
21:22.01Qwellyou can always svn ls
21:22.07mrgobyyou should be able to browse the repos via http as well
21:22.08_victormrgoby, http://pastebin.ca/33233
21:22.10Qwellsvn ls http://svn.digium.com/svn/zaptel/tags/
21:22.20_victori have put the sip.conf file and the sip debug output
21:22.35wshs"Reliably Transmitting: SIP/2.0 404 Not Found" what in the world does that mean?
21:22.53_victorthe sjphone software is configured with user sjphone, password blah
21:23.09_victorit should work (i think).. it works with SER with similar settings
21:23.15_victorbut it doesn't
21:23.31_victorinstead of replying with an authentication required, it replies with 404 not found
21:23.32Duttscheers Qwell.....
21:24.41Renacorhaving a queue setup like this http://pastebin.com/458809 will set it up so it will read out approx wait times instead of number in line won't it?
21:24.45mrgoby0357229001
21:24.55mrgobydoes that number look familiar to your , victor ?
21:25.10_victoryes, it does
21:25.18_victorit is configured as caller id in sjphone
21:25.44mrgobyok, try setting that to your sjphone login name... you are sending that as your sip username,
21:25.53mrgobyi believe
21:25.56*** join/#asterisk darby_t (n=tom@dmz49.neoplus.adsl.tpnet.pl)
21:26.02mrgobysjphone can be confusing to config
21:26.16_victorok, you mean like putting username=0357229001 in sip.conf?
21:27.01mrgobyyes, make a duplicate entry to your [sjphone] entry with  [0357229001] and see if it works
21:27.10mrgobyin sip.conf
21:27.25mrgobyand do a reload on asterisk to make it reload sip.conf
21:27.26shido6anyone own a honda civic?
21:27.35mrgobyshido6, yes
21:27.40shido6whats the firing order?
21:27.49mrgobyon the pistons ?
21:27.52shido6yes
21:28.02mrgobyi have no idea, that is why i bought a honda civic
21:28.06shido6LOL!
21:28.12shido6good answer.
21:28.12mrgoby:-D
21:28.27Renacorhaving a queue setup like this http://pastebin.com/458809 will set it up so it will read out approx wait times instead of number in line won't it?
21:28.30wshs"Looking for <my phone number> in broadvoice-inbound" thought i had it configured to go to exten 101, which i obviously didn't do correctly. so, next question, how do i partially cure my retardedness and make it go to exten 101?
21:29.37mrgobywshs: from where ?  somewhere in the dialplan, or from connection ?
21:29.51wshsfrom connection, in sip debug 'mode'
21:30.25wshsi could just as easily rename the extension in the context to the phone number, but that makes tracking of things in the long run more difficult
21:30.33mrgobyit goes to s by default, so you could have an entry   exten => s,1,goto(101)
21:31.09*** join/#asterisk Lurr (n=pr0ph3t@216.64.27.125)
21:31.23*** part/#asterisk Lurr (n=pr0ph3t@216.64.27.125)
21:31.24mrgobyi'm not sure what you are trying to accomplish though, that is just an ugly hack
21:31.24wshsgot register set to "<number>@sip.broadvoice.com:<pass>:<number>@sip.broadvoice.com/101".. shouldn't that /101 send it into extension 101?
21:32.27_victormrgoby, no luck.. can you indicate me a sip client you are familiar with for my initial tests? after i am sure the config is ok, i can try again with sjphone..
21:32.54mrgobyxlite works okay, it has been a long while since i have messed with soft clients, honestly
21:34.00wshsmrgoby, somehow, i managed to screw up something that affected incoming calls for broadvoice. set debug on, and found that message which sorta stood out like a sore thumb, indicating that the extension i thought i had configured isn't configured correctly.
21:34.16_victormrgoby, ok.. i don't want also to play with soft clients.. but the hard phones i need to play with are in the office, and now it is week-end ..
21:34.41_victormy final goal is to use asterisk as sip to h.323 translator for a video call between a sip and a h.323 video phone.. if i will be able to do this
21:35.18mrgobyerm, you are in for a world of hurt if you are looking to use h323
21:35.35mrgobyit can but shouldnt be done, imho
21:35.53*** join/#asterisk f0urtyfive (n=noone@71.225.226.175)
21:36.48_victori failed to find any other good h.323 to sip proxy.. and i have to call from a sip videophone to a h.323 to isdn video gateway (no sip to isdn video gateways existing currently) and from there to an isdn video phone..
21:37.30_victori don't really want to use asterisk.. i was pretty happy until yesterday with my cirpack ss7&sip switch (but no video on it unfortunately and that's what i have to do now)
21:41.44lofi-revis this the correct syntax?:  exten =>  _NXXNXXXXXX,2,Dial(IAX2/persona|30|t&IAX2/personb|30|t)
21:47.07marc32422i have calls coming from ser to * for voicemail. How do I rewrite so that all other packets goes through ser, and not *
21:47.33marc32422once voicemail is done.
21:50.29*** join/#asterisk f0urtyfive (n=noone@71.225.226.175)
21:51.10*** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net)
21:52.15*** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
21:59.14EriSanare there any sip/iax windows clients that have also video ?
22:01.53f0urtyfivegah can anyone lend a hand with an insane Cisco 7960
22:02.26*** join/#asterisk jebba (n=jebba@200.115.209.178)
22:02.26robl^try a sledgehammer!  :)
22:02.34Nuggetheh
22:03.59f0urtyfivebasically, I can ping it inside the network I can ping it outside the network, and my other sip phone can connect to the server and make calls
22:04.11jake1932before you do that - i'll send you a box with a pre-printed shipping label
22:04.31f0urtyfivethis one says "Network delay, Trying backup" but no packets ever get to the server (packet sniffing server side)
22:04.43jebbai have an iaxy. When I try to configure it with iaxyprov,  a'la   `iaxyprov 192.168.200.235 iaxyba1.conf`    it just returns immediately (it says nothing--no errors, no "ok", no nothing), and I just get a prompt. It doesn't appear to be configuring the iaxy.  Ideas? :))
22:04.44f0urtyfiveand, it was working a few days ago
22:09.02wshswhen i have _X11-like numbers configured, how do i make asterisk dial them immediately instead of waiting for input timeout?
22:10.47file[laptop]wshs: is this on a SIP device?
22:11.52wshsi have the sip device configured such that asterisk gets the input instead of waiting for the input and then passing onto asterisk
22:13.07file[laptop]so you're using early dial?
22:14.17*** part/#asterisk ctooley (n=ctooley@jc1-111.moment.net)
22:14.25wshsif that's what it's called, sure. (have asterisk generating the tones and waiting for input)
22:14.32file[laptop]oh DISA?
22:15.05tzangerwhoa
22:15.06mishehuI know this has been discussed a lot in the past, but I'm not sure what keywords to search on.  suppose I want to change the CID for a call but not have it also change the cdr for hte call, what should I search on (or if you know an exact URL for me to read, let me know)
22:15.06marc32422how do you connect the dlink des-1024D to the wan?
22:15.07tzangern-trance?
22:15.08tzangerhahaha
22:15.48wshsfile[laptop], im not to keen on terms, just what things do. still learning, and still have a ton of learning to do.
22:16.07wshsah
22:16.09wshsDISA it is
22:17.08twistedyay
22:17.10wshsalthough im not using the disa command
22:17.12twistedtoday is moving shit in the apartment day
22:17.51twistedhi file[laptop] :)
22:18.26_Thorhello everyone
22:18.35file[laptop]twisted: do I dare inquire why you're moving stuff?
22:18.41twistedi'm just re-arranging
22:18.46file[laptop]uh oh
22:18.47file[laptop]that can't end well
22:18.51_Thoranybody: where is mysql.h?
22:18.52twistedmoving my desk into the main room from the bedroom
22:18.57twistedsince i'm never in the bedroom
22:19.14file[laptop]_Thor: that's a complicated question, and since you're probably asking it, do you have the MySQL client development stuff installed?
22:19.21file[laptop]twisted: ah
22:19.24twistedmoving my machine from the coffee table to the desk, bringing in a chest of drawers for the bedroom, etc.
22:20.07_Thornoo
22:20.26twistedgirls don't dig big computer setups in the bedroom :P
22:20.51mishehuno they don't unless they're ubergeekettes
22:20.55twistedtrue
22:21.09_ThorI downloaded the asterisk-addons and I can't get to compile the cdr_mysql yet
22:21.09twistedbut the chances of me finding one of those where I live is slim to none :P
22:21.18file[laptop]'tsk 'tsk
22:21.22file[laptop]aren't you optimistic
22:21.31twistedfile[laptop], i've looked.
22:21.55*** join/#asterisk CRCC (n=crc@dslgw1.astra-net.com)
22:22.08_ThorIt says it's missing mysql.h and errmsg.h, and mysql_version.h
22:22.17CRCCanyone using orderlycalls
22:22.37mishehu_Thor: and you do have mysql installed right?
22:22.40twistedbut the joys of wireless lets me take my machines anywhere in here without cabling mess :)
22:22.50mishehunot to ignore the obvious question
22:22.50mishehuheh
22:23.02twisted_Thor, you need the mysql-devel libs
22:23.08*** join/#asterisk Zach^^ (n=Zachary@65.121.244.130)
22:23.14Qwelltwisted: Got a second for a msg?
22:23.16_Thorok, thank you gys
22:23.20Zach^^i am trying to configure my Grandstream 2000
22:23.29twistedQwell, maybe..
22:24.01Zach^^and when i go to http://192.168.0.107/config_a1.htm (that is the ip of the phone) i get You are not authorized to view this page
22:24.04_Thormisheshu: thank you
22:24.48*** part/#asterisk jebba (n=jebba@200.115.209.178)
22:24.48twistedZach^^, http://192.168.0.107/
22:24.51twisteddon't go to config_a1
22:24.55twistedlet the phone take you
22:24.58Zach^^i did
22:25.04Zach^^and it takes me to that page
22:25.36Zach^^when i go to adv settings
22:26.28Zach^^Product Model:    GXP2000
22:26.28Zach^^Software Version:    Program-- 1.0.1.9    Bootloader-- 1.0.1.2
22:28.43*** join/#asterisk CyberPony (n=CyberPon@cpe-069-132-017-022.carolina.res.rr.com)
22:29.00lofi-revhow do I combine a timeout and simultaneous ringing into a single Dial command?
22:29.02Zach^^status and basic settings are they only to that work
22:29.21*** join/#asterisk CRCC (n=crc@sharpstrong.proxy.astra-net.com)
22:29.48CRCCneed help with orderlycalls
22:30.15Zach^^can anyone help with the grandstream?
22:30.20CRCCin some way a get extra space when send command to asterisk
22:30.49CRCCexample call.setextension(3000)
22:31.02CRCCasterisk receives '3000 '
22:31.44CRCCasterisk 1.2.1 suse 10 ordelycalls on win mashine
22:31.58*** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
22:31.58*** mode/#asterisk [+o twisted[mobile]] by ChanServ
22:32.11Qwelloh great, he's mobile...now we're all doomed
22:32.16twisted[mobile]lol
22:32.19QwellI mean...Hi twisted!
22:32.24denonhaha
22:32.55twisted[mobile]i'm now tearing apart my desktop setup to move it to the desk that i just moved into the main room :P
22:33.14file[laptop]-right-
22:34.10Zach^^anyone have that problem with the GX2000
22:34.31Qwell[hC]: still around?
22:35.32wshsfile[laptop], had some ugly stuff going on to duplicate disa, because i didn't know that command existed. got rid of that. so, any way to make asterisk immediately process X11 numbers instead of waiting?
22:36.42*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:38.56tzangerok last.fm fucking rocks.
22:38.58tzangerthat is all.
22:39.28Nuggetheh
22:39.34NuggetI'm "MacNugget" on last.fm
22:39.44Nuggethttp://www.last.fm/user/MacNugget/
22:39.49*** join/#asterisk moy (n=kvirc@dsl-201-133-179-47.prod-infinitum.com.mx)
22:41.25tzangerheh I'm tzanger, oddly enough
22:41.29Zach^^anyone here have a grandstream?
22:42.27tzangerwow you use this quite a bit
22:42.32tzangerI'm doing it all thorugh the proxy client
22:42.45*** join/#asterisk degadar (n=grahamty@217.14.136.6)
22:42.48tzangerso I can listen to it on the headless box connected to the receiver
22:43.07NuggetI loaded the client a long time ago and I just never think about it now
22:43.35NuggetI never actually listen to music through them, it just tracks what my iTunes is doing
22:43.43tzangerahh
22:43.49shido6ok you guys are gonna make me reboot
22:43.52shido6Im feeling macjealous
22:43.57tzanger?
22:44.01tzangerreboot for why?
22:44.07shido6reboot into osx :)
22:44.19tzangerI could see hacking up the proxy so I can have better control but so far it's good
22:44.29tzangershido6: :-)  I have an osx-x86 pc here too it hasn't been turned on in a while
22:45.30tzangerwhy do you subscribe to it if you don't use it
22:47.02NuggetI like the concept and I want to support the site
22:47.21denonhmm, isnt there a 1.2.1 svn branch?
22:47.25tzangerah
22:48.57*** join/#asterisk darby_t (n=tom@dmz49.neoplus.adsl.tpnet.pl)
22:49.08Qwelldenon: tag
22:49.19Qwell1.2 branch, 1.2.x tags
22:49.24denonah, duh
22:49.29denonI just noticed that heh
22:49.30*** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
22:49.30*** mode/#asterisk [+o twisted] by ChanServ
22:49.33twisted[mobile]incoming!
22:49.34twisted[mobile]er
22:49.36twisted[mobile]here
22:49.38*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:49.39*** mode/#asterisk [+o drumkilla] by ChanServ
22:49.42Qwellack!
22:50.02drumkillasyn?
22:52.20drumkillalots going on in here ...
22:52.24ManxPowersynack?
22:52.26*** part/#asterisk degadar (n=grahamty@217.14.136.6)
22:53.05mrgobywhat does 'call failed to go through, reason 0' indicate, coming from pbx_spool ... ?   it is produced when i use a callfile with iax...
22:53.16*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
22:54.31*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:54.40*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
22:54.49AbbasManxPower:  i am using Chanunavail between 2 termination GWs     if one has no channel call goes on next termination GW but in CDR  disposition comes FAILED for that call which was completed by the next GW due to chanunavail
22:55.08shido6oooh
22:56.00mrgobythat sounds like danger
22:56.35*** join/#asterisk Vendetta9 (i=Vendetta@209-112-190-184-cdsl-rb1.anc.acsalaska.net)
22:56.41QwellNugget, tzanger: hope you don't mind my adding you guys :P
22:56.54tzangerQwell: not at all... I didn't even have the decency to ask nugget if he minded.  :-)
22:57.07Qwellnor he you :P
22:57.20tzangerhahaha
22:57.24QwellIf he did...dunno
22:57.31twistedwheeee!
22:57.45tzangerfuck I love last.fm...  I am gonna give it a couple weeks to see if the proxy really does work and if so, my US$36 for the year is off
22:57.48Qwellahh, he did.  heh
22:57.53twisted[mobile]hmm
22:58.01twisted[mobile]last bit of moving includes sound :P
22:58.03Nuggetheh
23:01.06tzangerI'd like to know if the stuttering is because they're running low on bandwidth or the connection to htem just sucks
23:02.23*** join/#asterisk docelmo (n=docE@static-71-251-95-3.tampfl.fios.verizon.net)
23:02.41QwellI'm on there as northantara, if you guys care :p
23:03.07docelmosay does anyone know how I can match on a call with no callerid using exten => 8135551212/???,1,...      What would I put under the ???
23:03.56Qwelldocelmo: I don't know, but I think there was a thing on the wiki
23:05.14drumkillaQwell: what are you guys talking about
23:05.29drumkillaI'm on "there" as drumkilla ... maybe ...
23:05.32drumkillaif I knew what it was :)
23:15.27ManxPowerdocelmo, exten => 5551515/,1,Noop(No CallerID") or exten => 5552323,1,GotoIf($[${LEN(${CALLERID})} = 0]?9:4)
23:16.58*** join/#asterisk zotz (n=zotz@24.231.47.168)
23:21.12AbbasManxPower:  i am using Chanunavail between 2 termination GWs     if one has no channel call goes on next termination GW but in CDR  disposition comes FAILED for that call which was completed by the next GW due to chanunavail
23:21.33ManxPowerAbbas, I cannot help you, I don't bill for calls and so never cared about the CDRs
23:21.41robl^is there a trick to using hint priorities with ZAP channels?  SIP works fine but something like exten => 5299,hint,ZAP/1 doesn't seem to work
23:24.03blitzragedrumkilla: !!!
23:24.18blitzragefile[laptop]: zup zup
23:24.54file[laptop]et tu?
23:25.13blitzrageI don't know enough french words yet :)
23:25.20file[laptop]ah
23:25.22file[laptop]"and you?"
23:25.27file[laptop]:P
23:25.36blitzragewell yah -- I figured that part out -- but couldn't reply :)
23:25.37blitzragelol
23:25.47file[laptop]'tsk 'tsk
23:26.36blitzragegod I hate pop up ads
23:31.30*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:31.36blitzrageit doesn't protect against all popups either (as I just attested to)
23:31.49tzangerheh
23:45.29blitzrageso, anyone here ride motorcycles?
23:45.47fugitivono, but i play guitar
23:45.55blitzrageheh... I play drums ;)
23:46.06fugitivogood, now we need a bass player
23:47.07Abbaswhat does ralaxdtmf does?
23:48.11blitzrageit gives the dtmf a beer and a comfy couch
23:48.24fugitivoand a remote control
23:50.16*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
23:52.39*** join/#asterisk Luhiwu (n=marsosa@200.63.89.241)
23:52.56CyberPonyI play bass and ride motorcycles.... just not at the same time :)
23:53.35Luhiwuhello all, i'm having problems with dtmf, i can transfer from one side but not from the other, and the dial command has tT on it.
23:54.38Luhiwuanyone knows where should i take a look?
23:57.18*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)

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