IRC log for #asterisk on 20080207

00:01.19*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:10.41*** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net)
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00:17.33*** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net)
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00:26.04*** join/#asterisk AndyGraybeal (n=mind@casanueva.wifi.frognet.net)
00:28.47*** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com)
00:30.00xp_prgwhat does asterisk gateway mean, what is a gateway too exactly?
00:30.42shmaltz~wiki gateway
00:31.06shmaltz~wiki Gateway (telecommunications)
00:33.39shmaltz~quiet
00:33.40jbotACTION ok, ok, I will be quiet. When you start making sense, that is.
00:34.31*** join/#asterisk angryuser (i=nononon@df01t2-212-195-117-162.d4.club-internet.fr)
00:36.34justdaveanyone know if the directories defined in /etc/asterisk.conf are available as variables in dialplans?
00:40.02[hC]If asterisk accepts a g729 call, and ends up going to someones voicemail, do i need a g729 license/codec for the message to be left in .g729 format?
00:42.16shmaltzjustdave, I don't think so, but for what purpose?
00:42.35justdaveit'd be a one-time setting, I can put it in [globals] easily enough I suppose :)
00:42.40shmaltz[hC], you need it just for accepting it
00:43.01justdaveI'm running a System() command to move an audio file on the filesystem after a user records it
00:43.29shmaltzjustdave, you could use the app_record and specify where it's stored
00:43.52justdaveI am specifying where it's stored. :)  In /tmp so it doesn't overwrite an existing file
00:44.08justdaveafter the user okays that they like what they recorded, then I want to move it overtop of the original
00:44.23justdavebut that means figuring out where Asterisk is looking for it, and I hate hard-coding pathnames in the dialplan
00:44.28shmaltzjustdave, http://pastebin.ca/894519
00:44.56shmaltzoh, so I guess globals will do
00:46.23justdaveUbuntu uses /usr/lib/asterisk/sounds, RHEL uses /var/lib/asterisk/sounds, want to be able to use the same script on both boxes
00:47.22justdaves/RHEL/source install/
00:47.32justdavehah, nice bot
00:47.41*** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net)
00:50.10[hC]shmaltz: no you dont. you dont need the codec to do passthru, just the format.
00:50.31shmaltz[hC], then you do need it for VM
00:51.08[TK]D-Fenderjustdave, make a constant yourself for each popular option and uncomment the one applicable
00:52.49justdavehe only needs it for voicemail if the user doesn't have a g.729-capable phone when he checks it
00:53.02[hC]the user does.
00:53.05justdaveonly need the license to transcaode it I believe
00:53.14[hC]I asked if app_voicemail would save in native g729
00:53.18[hC]and i think it does.
00:53.38sbingnerright
00:53.48sbingnerto justdave
00:53.50*** join/#asterisk weazahl_ (n=jeremy@adsl-68-90-168-7.dsl.ksc2mo.swbell.net)
00:54.07[hC]yes that is correct, I just didnt know if app_voicemail transcoded it to g729 to save it or not
00:54.08[hC]I'll just check.,
00:54.09sbingneryou need to codec to play the prompts to the user on g729, unless perhaps if you have them all recorded in g729 already
00:56.00drmessanooh crap
00:56.11drmessanos/oh crap/gee whiz/
00:56.56weazahl_what a day, DSL went down for the entire city last night.  9pm to 5am.  i was in withdraw...  then this morning i had to deal with the aftermath.  of course, 1 of the 2 sites that didnt comeback up was my shop.
00:57.06sbingners/g729/evil proprietary expensive format/
00:57.17sbingnerwth it only did it once?
00:57.37sbingners/it/test/g
00:57.42weazahl_so we had one analogue line to run the store on this morning.  fun
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01:25.10husimonso how much echo cancellation do you folks usually deploy?
01:28.56*** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-217-198-130.nsw.bigpond.net.au)
01:29.49[TK]D-Fenderhusimon, at leasty thirty.
01:31.53husimon[TK]D-Fender, What if you have hardware with EC built in, does it just automatically try and remove echo?  Or is that dependent on what hardware you have.
01:32.12*** join/#asterisk adjohn (n=adjohn@219.106.248.145)
01:32.20[TK]D-Fenderhusimon, if it detects echo it will activate and suppress it
01:33.22Kobazhmm, this isn't good
01:33.28husimon[TK]D-Fender, hmm yeah I have EC hardware but still a slight bit of echol, so I'm thinking about enabling EC in *.  I could turn off the hardware EC and only do *, or try both.  Which do you think is the better route?
01:33.32Kobazlatest asterisk stable seg faults on startup
01:34.13[TK]D-Fenderhusimon, perhaps you should tell us what you have, show us how you are using it (and that it is in use), and show some care in the question you ask.
01:34.40husimon[TK]D-Fender, well the hardware appears to have no way to controlling the actual EC, except for on off, that's why I didn't give you further information.
01:35.05husimoni'm not using any EC in * yet.
01:35.09[TK]D-Fenderhusimon, including the MAKE and MODEL.  Yes, very informative...
01:35.41husimon[TK]D-Fender, sure i'll tell you but you won't like it ;)  redfone fonebridge2-ec 2 port pri.  Uses tdmoe (yes I know you hate this)
01:35.43b11dKobaz.. works for me.
01:35.50Kobazyeah
01:35.51Kobazheh
01:35.52Kobazthat's good
01:35.59b11daye :)
01:36.23[TK]D-Fenderhusimon, yeah...
01:36.26[TK]D-Fender~wglwat
01:36.26jbotit has been said that wglwat is well, good luck with all that
01:36.47[TK]D-FenderCrap-tastic
01:37.10b11dhusimon.. where are you getting echo anyways? just between your voip phones and the PRI?
01:37.27husimonecho between voip phones and outside numbers
01:37.33husimonnot major
01:37.37husimonbut sometimes I can hear bits of myself talking
01:37.38Kobazwhat's weird
01:37.46Kobazis if i spawn asterisk as a daemon, it doesn't seg fault
01:37.52Kobazif i start it in the foreground, it crashes
01:37.58b11dand when you attach to it afterwards?
01:38.01husimonkobaz what does the log say?
01:38.21b11dhusimon.. you're just crazy.. its all in your head.
01:38.22Kobazsome undefined symbols for the queue and meetme modules
01:38.25husimonblld laugh
01:38.31husimonblld i've had users mention it too
01:38.35b11dheh
01:38.53b11ddo you have echotraining enabled in your zapata.conf ?
01:39.05Kobazhmm
01:39.08Kobazi think so
01:39.24b11dim asking husimon.. not you Kobaz :)
01:39.34Kobazoh
01:39.36b11dsorry
01:39.39[TK]D-Fenderb11d, shouldn't have to.
01:39.50b11ddoes it segfault when you attach to the daemon Kobaz?
01:39.51husimonyeah i don't blld
01:39.59b11dhmm.. you enabled it for me TK :)
01:40.07b11dhusimon.. try it?
01:41.01husimonis that for * ec or hardware ec
01:41.07husimonatm I only have hardware ec
01:41.14[TK]D-Fenderb11d, yes, but in his case those vars can't control the remote TDMoE interface.  it delivers pre-processed frames.
01:41.14husimonenabled that is.
01:41.20b11doh :)
01:41.32husimoni'm emailing the company to askt hem
01:41.33husimonask them
01:50.04*** part/#asterisk beek (n=klinebl@65.211.106.243)
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01:52.34b11dttyl al
01:52.35b11dall
01:55.45lunaphytei guess the dialplan is not applied to dialed extensions in a linear fashion, from top to bottom?
01:56.46husimoni think so
01:56.51husimonexcept for the priorities
01:56.54husimonwhich it goes by number
01:58.15lunaphyteoh, oops.  looks like you're right.  i had a line unplugged.
01:59.35lunaphyteso one could craft a dialplan with overlapping patterns to try 1 action first and then another if not successful?
02:00.51husimonwhat do you men?
02:00.56husimontry one outbound trunk then another?
02:01.06lunaphyteyeah, more or less.
02:01.33lunaphyteremember my questions from yesterday, about a long list of prefixes?
02:02.25lunaphytethe point with all of that is that not all of the prefixes in my area code are considered "local"
02:02.43*** join/#asterisk adjohn (n=adjohn@219.106.248.145)
02:03.10lunaphyteso my goal is to send all calls to truly local pots prefixes out the pots line, and then send any other prefixes out through my itsp.
02:03.34lunaphyteand, at the same time, not require 10 digit dialing for those non-local prefixes.
02:03.51husimonso shouldn't that be a simple matter of listing all the local calls as patterns
02:05.15lunaphyteyeah, i was able to do that, with your guys' help - but, to match the remainder means i have to use a pattern that overlaps with the local patterns.
02:05.26lunaphyteat least, that was the only way i could figure to do it.
02:05.40*** join/#asterisk spyder12345 (n=kyle@rrcs-67-78-17-78.se.biz.rr.com)
02:06.11*** join/#asterisk HeXeD (n=hex@87-194-8-43.bethere.co.uk)
02:06.36lunaphyteso i was a little concerned that when a local pattern was dialed, it might get picked up by the "catch-all" pattern and get sent out through the itsp instead of pots.
02:06.51spyder12345With version 1.6 how can I play a sip call using tcp?
02:07.33lunaphyteit seems to be working, though i wonder if there might be a better approach.
02:08.03husimonlunaphyte, how come there is overlap?
02:08.11husimonshouldn't the area code decide if it is local or not
02:08.52lunaphytethe idea was to eliminate the need to dial the area code.
02:09.33husimonlunaphyte, so just send all 7 digit numbers to your pots
02:09.54husimonthen create patterns that handle the local area codes and send those to the pots
02:10.13husimoni think I understand your question now
02:10.20husimonbecause after those 2 rules you require a catch all
02:10.23phixok I am here again to try and get my ports working
02:10.23husimonright?
02:10.39phixso I will try to compile zaptel and the zaptel modules from source and see how that goes
02:10.44phixany other ideas?
02:11.35[TK]D-Fenderlunaphyte : there is, stop using catch-alls
02:12.02phix[TK]D-Fender: hey! :)
02:12.06husimon[TK]D-Fender, how do you then tell all area codes besides a few to go through a given trunk?
02:12.09[TK]D-Fenderlunaphyte : or use one that checks if it would match a specific series you'd route differently and branch off as needed
02:12.19phix[TK]D-Fender: How are you with TDM400p's?
02:12.23lunaphytehusimon: if i send all 7 digit numbers to pots, some will have errors, because even though they're in the same area code, certain prefixes are not local and require 1 + the area code be included.
02:12.28[TK]D-Fenderphix, in what way?
02:12.51husimonlunaphyte, that's your users problem for not dialing in the correct way then.
02:12.54[TK]D-Fenderlunaphyte : ok, whats the problem with this?
02:14.00phix[TK]D-Fender: I am trying to figure out why I can only get 2 out of 3 FXS working
02:14.10*** join/#asterisk jwh (n=jwh@kelley.ber.rewt.org.uk)
02:14.29phix[TK]D-Fender: working as in I have a dial tone, the third one sounds like zaptel or asterisk isn't running (no dial tone, amps everything sent to it)
02:14.59[TK]D-Fenderphix, be very careful because the ports in the back aren't in the order you might think
02:15.01phixthe zaptel module (wctdm) picks up all three ports
02:15.08lunaphytelet me back up a bit and try to state my goal.
02:15.39[TK]D-Fenderlunaphyte : Go deal with "1NXXXXXX"
02:15.50phix[TK]D-Fender: oh ok, when looking towards the back plane, right to left is ports 1, 2, 3, 4?
02:16.16phixor looking left to right, 4, 3, 2, 1 even
02:16.16phix:)
02:18.28spyder12345Does anyone know with version 1.6 when dialing via sip how you specify tcp instead of udp?
02:18.56[hC]phix: try one. if it doesnt work, try the other.
02:18.56[hC]:)
02:20.16*** join/#asterisk jpeeler (n=jpeeler@adsl-249-75-145.hsv.bellsouth.net)
02:22.42lunaphytein my area code, prefixes are split into 2 groups - local and "short-distance" (or whatever it's called).  local calls can be dialed with 7 digits, but short-distance calls must require 10 digit dialing.  my pots line has only local service on it, so no short-distance or long-distance calls can be made on it
02:22.58*** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net)
02:23.10phix[hC]: :) only two work out of three!
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02:23.39lunaphyteadditionally, it would be great to not have to remember or lookup, or do trial and error each time a calls is made, which prefixes require 10 digit dialing and which don't.
02:24.46[TK]D-Fenderphix, the modules on the card are something like 4,1,2,3  IIRC
02:25.02husimonlunaphyte, i dunno seems like you can't do that because the numbers will overlap
02:25.22husimonlunaphyte, the solution is probably to just require people use 10 digit for all numbers.
02:25.55lunaphyteultimately, if that's the case, that's ok.  i'm just experimenting w/ * for the time being.
02:26.11husimontk might have better ideas on how to solve that
02:26.23husimonthat was only my first reaction
02:30.09lmadsenlunaphyte: 'phyte' is what my old bbs was called
02:30.21lmadsenactually... it was 'phyte!'
02:30.21lmadsenheh
02:30.36lunaphytefunny :)
02:30.54*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
02:31.27lmadsenone of my biggest regrets was formatting the HD of that computer without backing up the bbs
02:31.49lmadsenit was on purpose because I'd had enough of the bbs, but would be awesome to have a copy of it for nostalgic value
02:32.11lunaphytesuch is hindsight...
02:32.22lmadsenindeed
02:33.03husimonhad enough of the bbs? what did it do to you?
02:33.04husimonheh
02:33.30lmadsennothing... I upgraded to internet :)
02:33.34lmadsenthe internets!
02:33.39lmadsena series of tubes
02:35.15lunaphyteyou never know, you could have had the next isca.
02:35.22alrslmadsen: there are still g-files around that list me as a cosysop of a board that hasn't existed for 18 years
02:36.03*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)
02:37.00lmadsenalrs: :D
02:37.07lmadsenya... the internet is a funny thing
02:37.24*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
02:37.29SwKanyone know a good source for refurb'd polycoms?
02:37.37ZaVoidnope :(
02:37.38lunaphyteme!
02:38.00lunaphyteoh, you probably mean phones... :(
02:38.04SwKyeah
02:38.07SwKi do
02:38.21lmadsenthat business exists already?
02:38.28lunaphytedarn, i've got a viewstation i'd like to unload.
02:38.37lmadsenlast time I heard about refurb'd phones, it was for a meridian system
02:38.58lmadsenanyone have an idea what I can use my Zaurus SL-5500 for? :)
02:39.13[TK]D-Fenderpaper-weight ;)
02:39.16lmadsenaye
02:40.53outtoluncprobably be useful as a remote control
02:41.52outtoluncsend it to me i will find a use for it <G>
02:41.53lunaphyteif you had something fairly lightweight nearby that you wanted to damage without having to get up, it could be handy for that.
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02:46.25*** join/#asterisk NoRemorse (n=fred@eth2459.vic.adsl.internode.on.net)
02:46.38NoRemorsehi all
02:46.39NoRemorsecan anyone tell me why if I directly edit sip.conf in trixbox and add a peer, it does not show up under the FreePBX
02:47.05lmadsenNoRemorse: see #freepbx for support
02:47.11ZaVoidbecause trixbox loads everything into a db i believe
02:47.15ZaVoidand what lmadsen said
02:47.21ZaVoidhey lmadsen question for you
02:47.28outtoluncshouldn't you be editing sip_additional/custom.conf something like thta.. and asking on #trixbox?
02:47.28lmadsenor #trixbox
02:47.43ZaVoidexten => s,n,Set(CHANNEL(language)=en)    <--- can i replace "en" with a variable i pull from my db via agi?
02:47.51lmadsenZaVoid: of course
02:47.53kyronlmadsen, and you could probably have ran your BBS off a telnet port like my buddy Synoptic does ;)
02:47.53Nuggettelnet is eeeeeeevil!
02:47.58*** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net)
02:48.14ZaVoidso exten => s,n,Set(CHANNEL(language)=${ARG3})
02:48.17ZaVoidis that right?
02:48.19lunaphyteyay telnet!
02:48.51ZaVoidor am i formatting that wrong?
02:48.53kyronNugget, nooooooooooo, it's fun for password sniffing ;)
02:48.56lmadsenexten => s,n,Set(CHANNEL(language)=${ODBC_SQL(SELECT language FROM my_users WHERE username = '${ARG1}')})
02:49.15kyronI prooved my point about using hubs and not switches at my first internship ;)
02:49.24ZaVoidyeah but i'll already have the variable from a sql query i did earlier in the dialplan
02:49.36kyronlmadsen, "why does it take so long for the calls to be answered"
02:49.47lmadsenZaVoid: format is correct -- or replace it with another function where you get data from.  ${DB(family/var)}) could be another place
02:49.52NoRemorsesip show peers in asterisk console lists the peer tho,
02:49.55ZaVoidok thanks man
02:50.17ZaVoidof course it does NoRemorse you've added it to the asterisk
02:50.21[TK]D-FenderNoRemorse, FreePBX does not READ * config files, it generates (obliterates) them.
02:50.27lmadsenkyron: collisions are hawt
02:50.41[TK]D-FenderNoRemorse, and...
02:50.42[TK]D-Fender~freepbx
02:50.43jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
02:50.45NoRemorseinteresting, i wonder why a2billing uses flat files instead of writing into the database :(
02:51.06NoRemorseok thanks
02:51.09*** part/#asterisk NoRemorse (n=fred@eth2459.vic.adsl.internode.on.net)
02:51.53*** join/#asterisk PepOSX (n=angeldav@190.72.146.71)
02:52.51lmadsenI also heard trixbox has the worlds largest asterisk community... so i always wonder why trixbox users keep coming in here asking for help
02:53.56[TK]D-Fenderlmadsen, Same reason Lisa-Marie Presley & Michael Jackson broke up... irreconcilable similarities ;)
02:54.18kyronlmadsen, LOL
02:54.30kyronlmadsen, sniffing used to be sooo esy
02:54.50ZaVoidlmadsen: so in my php i'm doing $agi->set_variable("LANG", intval($row['lang'])); so i can just Set(CHANNEL(language)=${LANG})    makes sense right?
02:55.16ZaVoidgonna try it in a few mins.. just wanna make sure i got the theory down... think i do
02:55.16lmadsenyes
02:55.21ZaVoidcool thanks
02:59.31*** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca)
03:05.52*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
03:05.52*** mode/#asterisk [+o russellb] by ChanServ
03:11.19ZaVoidwhos bot records everything for ircarchive.info ?
03:11.47lmadsenjbot
03:11.58lmadsenjbot: who owns you?
03:11.59jbotTimRiker does
03:14.29lmadsenok... so I've a customer who needs me to answer a fax from a zap channel, then dial out another zap channel to a fax server. Once the fax server answers, I need to play some DTMF to the fax server. I see the D() option in Dial(), but it says, "Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged"....so I wonder if that will work or not.... anyone try the D() flag before?
03:15.15[TK]D-Fenderlmadsen, That'd work.
03:15.21lmadsenoh ya? ok swet
03:15.23lmadsensweet* :)
03:15.47lmadsenI'm just a bit confused by what "but before the call gets bridged"
03:15.50lmadsenmeans
03:16.21ZaVoidmaybe the rtp he means?
03:16.22[TK]D-Fenderlmadsen, means you pass on DTMF before the adio is bridged from the calling side to the called (you don't hear the DTMF or anything till after its done)
03:16.34lmadsenahhhhhhhh
03:16.38lmadsengokie
03:16.54lmadsencute quit msg :)
03:19.43JunK-Yhttp://www.monopolyworldvote.com/en_US/world/leaders  , wow montreal is #1!
03:20.27russellbheh, jpeeler must be trying to set up his tdm400p at home
03:20.51lmadsen:)
03:26.36*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
03:26.51lmadsenanyone wanna try faxing something to me? :)
03:27.10*** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye)
03:28.52lmadsen:D
03:29.01lmadsenthis is actually a customers fax line, so no worries :)
03:29.19_ShrikEI test with http://www.interpage.net/sub-wwwfax.html
03:29.34_ShrikEnot sure about calling canada though
03:30.11outtolunchttp://www.tpc.int/
03:30.12lmadsenthis is a US number
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03:30.27lmadsen_ShrikE: oh hawt... thx!
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03:38.53eric2anyone use callweaver?
03:38.56eric2for faxing
03:41.35ZaVoidwhy do people still fax? :(
03:42.17justdaveI just scan to a pdf and email it to the recipient :)
03:42.56ZaVoidremember that company visonx or somthing like that.. that back in mid 90's made personal scanners(single sheet) that sat behind your keyboard?
03:42.57coppicealthough it is stupid from a technical point of view, faxing has various legal implications
03:42.59*** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat)
03:43.22_ShrikEeric2: I have used callweaver with t38gateway with some pretty good success.
03:44.12eric2I need faxing for my customers
03:44.20eric2ya, that's right, legacy faxing
03:44.27ZaVoidoh snap they still around: http://orders.visioneer.com/category.jsp?category=MOBILE
03:44.34ZaVoidalthought they cost more now then in the 90's
03:44.35eric2_ShrikE  did you use a 3rd party for faxing?
03:44.50eric2got any recommendations?
03:45.13riddleboxwhat sites do you guys buy your phones from?
03:45.28*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
03:45.28ZaVoidyeah PAPERPORT http://support.visioneer.com/products/mobile/EOL/Sheetfed/default.asp
03:45.33ZaVoidlets talk oldschool :)
03:45.35_ShrikEjust get the latest spandsp and callweaver and give it a try.  I am using it with an audiocodes mp-114.
03:45.58eric2riddlebox - gentek.com
03:46.19ZaVoidi buy mine form abptech
03:47.33ZaVoidanyone else use this new program called skitch?  im in love with it
03:47.44eric2what does it do?
03:48.03*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
03:48.19*** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290)
03:48.51ZaVoidlets you take a snapshot of anyhwere on your screen and lets you easily mark it up.. type on it.. circle stuff etc etc etc then quick upload to a server of yours and gives you the  link
03:48.55L|NUXHello every one
03:48.59L|NUXcan some one tell me how can i brodcast livestream on sip
03:49.22eric2L|NUX - that's a good question
03:49.48L|NUXeric2: thanks but there should be a good answer for that question :)
03:50.10husimonyeah I really don't understand why faxing is around still
03:50.17[TK]D-FenderL|NUX, SIP is not a "broadcast" medium
03:50.29*** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net)
03:50.36[TK]D-Fenderhusimon, in a word : luddites
03:50.41*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
03:50.41*** mode/#asterisk [+o russellb] by ChanServ
03:50.58husimonlaugh
03:51.13eric2faxing is used for legal docs
03:51.20eric2main reason as far as I know
03:51.20L|NUX[TK]D-Fender: i can understand
03:51.20husimonit's not like you couldn't scan them
03:51.29eric2people are old skool
03:51.36L|NUX[TK]D-Fender : but can we play mms:// stream using some program on sip ?
03:51.45husimonLINUX EH!?
03:51.59L|NUX[TK]D-Fender : so that when some one call to number they can listen stream ?
03:52.22coppiceSarbanes Oxley has created a resurgence of faxing
03:53.12*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
03:53.12*** mode/#asterisk [+o russellb] by ChanServ
03:54.14*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)
03:55.18phixhi
03:55.30[TK]D-FenderL|NUX, use something like rawplayer for MoH and tune to a broadcast
03:56.24eric2callweaver is only used if you have additional hardware... no?
03:56.32riddlebox[TK]D-Fender, do you recomend using a TDM808B for a system that will have 7 lines? or should I get two 4 port cards?
03:57.01[TK]D-Fenderriddlebox, partial PRI too pricey?
03:57.11riddlebox[TK]D-Fender, yeah
03:57.23riddleboxand the company doesnt want to use it for some reason
03:57.40[TK]D-Fenderriddlebox, Only justifiable one is money.
03:57.45*** join/#asterisk adjohn (n=adjohn@219.106.248.145)
03:57.54riddleboxohh yeah they need to use a union company to do it and AT&T is the only union one around
03:57.59eric2what's the competitive price on a PRI these days?
03:58.10riddleboxeric2, it depends on your area
03:58.29eric2ah
03:58.39*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
03:59.32methodswhy when i type a number it keeps changing it to sip:<number>@mysipserver
03:59.38methods??? i just want it to call a land line
04:00.22[TK]D-Fendermethods, "it">
04:00.30[TK]D-Fender?
04:00.36methodsi tried 2 software app's
04:00.38methodsboth do the same thing
04:00.59[TK]D-Fendermethods, so, whats the actual impact of the fact it visually reformats the URI like that?
04:01.15methodswell i want it to call a land line
04:01.22methodsis it in fact doing that ?
04:01.38[TK]D-Fendermethods, You haven't shown us anything and we're not psychic.
04:01.48[TK]D-Fendermethods, PASTEBIN is your friend.
04:01.54[TK]D-Fendermethods, and so is SIP DEBUG
04:02.00methodsdude what in world do you want me to pastebin ?
04:02.12methodsi'm not running the asterisk box
04:02.15methodsi'm connecting to it
04:03.27[TK]D-Fendermethods, if we can't see the SIP communication then we can't tell whats happening now can we?
04:03.45[TK]D-Fendermethods, and if you don't have access to the server then you're really up a creek
04:04.08[TK]D-Fendermethods, it could be misconfigured and I guess you'll never know for sure how or why.
04:05.24methodsReceived from: 66.55.150.197:5060
04:05.24methodsSIP/2.0 404 Not Found
04:05.24methodsVia: SIP/2.0/UDP
04:05.38phix[TK]D-Fender: any ideas?
04:05.48[TK]D-Fendermethods, pastebin the whole thing please.  Many things can say 404
04:06.03[TK]D-Fenderphix, on?
04:06.05methodsthe number i tried to dial was 404
04:06.11methodsit looked like it had my password
04:06.23phix[TK]D-Fender: my TDM400p issue
04:06.29phix[TK]D-Fender: I have tried all ports
04:06.46phix(I even moved one from the thrid position (the one that wasn't working) to the 4th
04:06.47[TK]D-Fenderphix, pastebin your configs & dmesg
04:07.16phixhttp://rafb.net/p/y1l9ih85.html
04:07.25phixI will do dmesg now
04:08.02*** join/#asterisk Robba (n=rob@203.56.181.15)
04:08.03phixactually it is 6 lines, I could probably prv msg it to you if you like
04:08.37[TK]D-Fenderphix, nope
04:08.49phixhmmm ok
04:10.01phixhttp://rafb.net/p/HBBIhG95.html
04:10.35AJayMNis there a way to raise the volume level of g729? i have the phones cranked up but its so quiet compaired to u711
04:11.18RobbaHi guys, for some reason unknown to me, my boss wants me to configure our asterisk server to have to dial 0 to get to an outside line, what would be the most simple way to do this?
04:12.45phix[TK]D-Fender: everything look in order? I didn't forget any important directives?
04:13.26[TK]D-FenderAJayMN, Nope.
04:13.39[TK]D-Fenderphix, looks fine. you've tried all ports an only 2 work?
04:17.47*** join/#asterisk asr33 (n=asr33@dsl-207-112-72-48.tor.primus.ca)
04:22.32Kobazheh
04:22.44*** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com)
04:22.44Kobaz[TK]D-Fender: you're always here :P
04:23.24phix[TK]D-Fender: correct
04:24.19[TK]D-Fenderphix, Ok, tell you what, SWAP your modules in place to see if you have a dead one.
04:25.52lmadsenfile: you still fail
04:27.36[TK]D-FenderRobba, put a 0 prefix on the appropriate extensions in your dialplan.
04:27.49*** join/#asterisk micander (n=Michael_@ip70-181-134-119.sd.sd.cox.net)
04:29.10*** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net)
04:30.36SteveTotarois Tzafrir around ?
04:31.28SteveTotaroanyone know bristuff?
04:32.55*** join/#asterisk LakeSolon (n=blake@12-202-198-20.client.mchsi.com)
04:34.15*** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245)
04:37.54phix[TK]D-Fender: ok
04:40.40*** join/#asterisk ManxPower (n=manxpowe@251.sub-70-223-214.myvzw.com)
04:46.04SteveTotaronobody can help with bristuff?
04:46.54*** join/#asterisk erojasv (n=erojasv@201.240.80.159)
04:51.19[TK]D-FenderSteveTotaro, Europe is closd right now, come back in a few hours
04:54.58drmessanolol
04:55.21drmessanoFiber being cut... IT'S THE FINAL COUNTDOWN
04:56.49JTi wonder who's cutting all the submarine fibre in the middle east
04:56.56[TK]D-Fenderdrmessano, about a month or so ago I was inspired to play "Carrie" on piano, learning from the memory of having heard it last a long time ago.
04:57.10drmessanoniice
04:57.41drmessanoAccording to The Register, it's 3 Fiber cuts
04:57.47drmessano4th was a power fail
04:58.08drmessanoand 2 of them are a few km apart at most, likely from the same event
04:58.15drmessanoIran didn't lose connectivity
04:58.25JTit's up to 5 fibre cuts
04:58.36drmessanoand with traffic being rerouted, egypt is almost back up
04:58.37drmessanono
04:58.39drmessano3 cuts
04:58.52*** join/#asterisk asr33 (n=asr33@dsl-207-112-48-78.tor.primus.ca)
04:59.18drmessanohttp://www.theregister.co.uk/2008/02/07/cut_underseas_cable_conspiracies/
04:59.20drmessanoThere you go
04:59.55husimondrmessano, i saw 5
05:00.03drmessanoNope
05:00.05drmessanoRead the link
05:00.20drmessanoReports were blown out of proportion and just outright wrong in a few cases
05:00.33husimonstill pretty unsettling
05:00.57drmessanoCoincidence it looks like
05:01.06drmessano2 of the cables were from one single event
05:02.10husimonyeah
05:02.18husimoncrazy coicidence
05:02.42*** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83)
05:03.12drmessanoAccording to Becket, there's nothing unusual about the number of outages. There are about 100 cut cables every year, enough to keep a fleet of 25 cable repair ships fully occupied.
05:03.16husimonyeah i saw that
05:03.17drmessanoThat says it all, really
05:03.58husimonreading the stories that said 5 in two weeks in the that area did make me think wtf though
05:04.01*** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au)
05:04.05drmessanoWell
05:04.09husimonobviously that wasn't the case
05:04.12drmessanoWelcome to the liberal fox news media
05:04.19drmessanoI saw the same things
05:04.39husimonfox news can jump off a cliff
05:04.41husimoni hate them so much
05:05.09husimoni'd really like to see another major network run a story on all the fucked up shit they do
05:05.13husimoninstead of it just being on internet sites
05:05.33drmessanoI remember the plane that had a stuck landing gear above LAX.. They were flying in circles to burn the fuel out so they could crash land with a near empty tank....
05:05.51drmessanoCNN: Plane with 130 on board in trouble over LAX
05:06.01husimonthe fox news headlines are so messed up
05:06.04drmessanoFOX: 130 prepare to die in horrific LAX airline disaster
05:06.08JTliberal fox news media?
05:06.11JTwho's joking?
05:06.20husimonright wing is more like it but...
05:06.23JTfox news is like a commedy channel
05:06.37husimoni prefer to watch colbert and the daily show then any news channel
05:06.38JTi think we only get it on cable here so people can laugh at it
05:06.53husimonfox news is their #1 content provider
05:07.34husimoni hate fox news just about as much as I hate ann coulter
05:07.36drmessanoFox news represents more of modern news media than CNN does.. it's all about sensationalism
05:07.38husimonomfg that bitch needs to die
05:07.54drmessanoCNN has it's problems too..
05:07.58drmessanoBut Fox.. yeah
05:08.59JTfox takes a very republican right wing standpoint
05:09.12JTand it's definitely sensationalist rubbish
05:10.23*** join/#asterisk AndyGraybeal (n=andy@node254.38.251.72.1dial.com)
05:12.10*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
05:13.01drmessanoI really think Fox completely fakes their viewpoints
05:14.06drmessanoI think they do it as an attack on the rest of the media, purely for ratings and entertainment value.. Like "Not Necessarily The News" did on HBO in the 80s lol
05:14.21[TK]D-Fenderdrmessano, no, I'm sure they are that #&$^ed up an actually believe that garbage
05:14.52drmessanoThat's also impossible to comprehend though
05:15.04drmessanolol
05:15.12[TK]D-Fenderdrmessano, "Never underestimate the power the stupid people in large groups"
05:15.20drmessanoTrue
05:16.04coppiceall news is faked all the time, as a matter of policy. if they only lied some of the time, you'd easily tell the difference, so they distort everything
05:16.28drmessanoIt's impossible to report actual news and make consistant ratings from it
05:16.33drmessanoNo doubt
05:16.55coppiceif you've ever been involved in something that got into the news, you'll know the reporting beared only a passing resemblence to what you know actually happened
05:18.00drmessanoI was involved in a disaster situation a few years back that made national news.. it was fascinating what angles they were looking at in questions I was asked
05:18.45drmessanoI remember one 30 minute phone interview I had with a reporter where I don't think i've ever had to try harder to carefully pick out my words
05:19.15drmessanoI came to truly realize why people are so vague and generic at times.. heh
05:21.19coppice"An inmate at the local asylum got into the laundry, raped all the women, and escaped"
05:21.20coppice"That's awful. Those poor women. We need a headline that can really capture their plight........ Got it - 'Nut screws washers, and bolts'"
05:21.30drmessanoROFLL
05:23.13[TK]D-Fendercoppice, comedy GOLD
05:24.35drmessanoIf you really want true lulz, read Digg
05:24.42*** join/#asterisk SomethingISOdd (n=TestMast@S010600a0d1757bfb.cg.shawcable.net)
05:24.44drmessanoNow that... is good stuff
05:24.52SomethingISOddhello all anyone here use h323 with asterisk
05:25.07drmessano"Ron Paul endorses Asterisk 1.6!!!!"
05:28.24coppiceAsterisk 1.6. Cindy Crawford's choice
05:28.53drmessano"Digium defiantly releases Asterisk 1.6 beta"
05:30.02*** part/#asterisk asr33 (n=asr33@dsl-207-112-48-78.tor.primus.ca)
05:30.17drmessanoThe trick is, you have to be able to base it in fact.. at some level.. because all nutjobs have a traceable level of fact, however warped it may be
05:32.54drmessano"Microsoft admits severe flaws in Windows 2000, releases Service Pack 4"
05:33.10drmessanoUh.. yeah.. ok
05:33.23drmessanoI suppose.. ok.. yep
05:33.27jwhthere is a severe flaw in win2k
05:33.31jwhthey EoL'd it
05:33.38jwhthats the flaw :P
05:33.40drmessanolol
05:35.16drmessanoI remember thinking how cool it was 10 years ago that we had an AP Newswire in our buildings
05:35.28drmessanoNow AP is absolutely boring
05:35.33drmessanoToo much "fact"
05:35.52jwhlol
05:38.23*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
05:40.02BBHossanyone using 1.6 with dundi?  I seem to be having some trouble peering with someone on 1.4.17
05:40.22*** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au)
05:40.35*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
05:41.11*** join/#asterisk GameGamer43 (n=GameGame@cpe-74-65-38-18.rochester.res.rr.com)
05:41.46GameGamer43has anyone ever seen asterisk pass a call from zap to the phone, but when the phone picks up asterisk doesnt acknowledge it and continues to ring
05:44.02*** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob)
05:44.05[TK]D-FenderGameGamer43, what is the "phone"?
05:44.24[TK]D-FenderGameGamer43, and its be a good idea to pastebin the entire attempt with all appropriate debugs enabled
05:45.24JTSteveTotaro: i know bristuff
05:46.02GameGamer43its not my issue neccessarily, but using cisco phone and tried a softphone
05:46.02GameGamer43using tdm400p
05:46.02GameGamer43with 4 fxo
05:46.36*** join/#asterisk Malkut (n=Malkut@rrcs-76-79-244-73.west.biz.rr.com)
05:46.44Malkuti am here
05:46.46GameGamer43this issue I described is Malkut's issue
05:47.08Malkutand what a bad issue it is
05:47.54GameGamer43Malkut: please pastebin the cli output of the issue
05:48.04Malkutwhich portion would u like?
05:48.36GameGamer43from the call coming in to the disconnect
05:48.43Malkuthttp://pastebin.ca/index.php
05:49.09GameGamer43need your pastebin link
05:49.23MalkutGameGamer43,  http://pastebin.ca/894730
05:50.50Malkutyou see it pickup the call and do this and that when the call is not really picked up
05:50.51*** join/#asterisk ravichandran (n=Ravi@ip72-206-113-190.om.om.cox.net)
05:51.32ravichandranany helpful soul around here
05:51.41Malkutravichandran,  get in line :)
05:52.03ravichandranI am facing a one way audio problem with my workflow and have been crying without help :(
05:52.13GameGamer43what way
05:52.15ravichandranone way audi with asterisk
05:52.18ravichandranaudio
05:52.21ravichandranhere is the call flow
05:53.06ravichandranthe call comes into the PSTN and from that it hits a switch called CopperCom that has an IPM blade on it. This switch is behind the firewall. The SIP trun is inside the IPM blade.
05:53.10ravichandrantrun=trunk
05:53.28ravichandranthe call then goes to Jasomie which is on the public IP (session border controller)
05:53.46ravichandranfrom here the call gets to Cisco Pix firewall and then into our asterisk server.
05:54.03drmessanowow
05:54.10ravichandranOur asterisk server plays back an IVR. when the user presses 5 it makes an outbound call using the outbound context
05:54.20GameGamer43theres more
05:54.23JTyeah cisco pix tend to screw sip
05:54.26JTjunk
05:54.28ravichandran:)
05:54.33drmessanoYep
05:54.52J4k3*bing*
05:54.53J4k3;)
05:55.07J4k3(j/k)
05:55.09ravichandranhow do I disable that pix for the time being to make sure that everything goes through that firewall into the server
05:55.26drmessanoWe've almost completely done away with Cisco VPN stuff
05:55.29drmessanoThank goodness
05:55.31GameGamer43the simple solution is a bonfire
05:55.40GameGamer43just throw the cisco pix in
05:55.49*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
05:55.58[TK]D-FenderPIX = real trouble for SIP
05:56.13ravichandranso how do I proceed from here
05:56.27coppiceSIP = real trouble for everyone
05:56.39BBHosshey anyone here using the DUNDi-test network or using DUNDi period?  There aren't any good docs for it
05:56.57GameGamer43[TK]D-Fender: http://pastebin.ca/894730 is the pastebin you asked for and all questions should go to Malkut
05:57.15GameGamer43BBHoss: I found one good doc for dundi some time back, let me see if I can dig it up
05:57.31Malkut<<<<
05:57.45ravichandrancan I disable the cisco pix for right now to just act like a dummy guy for right now
05:58.03BBHossGameGamer43, thanks, i need a working doc because I believe i'm experiencing a bug in 1.6 that needs to be reported
05:58.31ravichandranif I can any leading tips would be appreciated
05:58.42GameGamer43BBHoss: what is your setup and are you trying load balancing or no
05:58.55[TK]D-FenderGameGamer43, [s@macro-dial:10] Dial("Zap/1-1", "SIP/200|15|tr") in new stack <- "r" = evil
05:58.56[TK]D-Fenderand
05:58.58[TK]D-Fender~freepbx
05:58.59jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
05:59.01BBHossno, i'm trying to join a peering network, but i can't get it to work
05:59.22ravichandrandrmessano are u there
05:59.23GameGamer43BBHoss: you look at this http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/
05:59.25drmessanoyes
05:59.34MalkutGameGamer43,  :(
05:59.45GameGamer43BBHoss: that one of the better guide I've ever found
05:59.56GameGamer43Malkut: i didnt forget about your issue yet
05:59.58GameGamer43lol
06:00.47Robbadoes anyone know if its possible to remove the need to press dial on a linksys SPA-941 to make a call?
06:01.06drmessanoProper dialplan
06:01.16Robbaproper?
06:01.17*** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net)
06:01.42drmessanoWhats your dialplan on the phones now?
06:02.35Robba10 and 8 digit dialling, and now that you mention it, it doesn't seem to dial after typing the 10 digits.... hmmm back to the drawing board
06:03.07drmessanoWhy not just load something generic?
06:03.12drmessanoLet Asterisk do the rest
06:05.44[TK]D-Fenderonly reason to need send is a bad phone dialplan
06:07.25drmessano(*x.|x.)
06:07.36drmessanoand let Asterisk do the real work
06:14.24*** join/#asterisk AJayMN (n=mypocket@76.201.155.119)
06:19.25AJayMNI upgraded to 1.2.26.1 last night and now my wakeup and weather isnt working anymore.. what could have been broke?
06:21.49*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
06:21.54[TK]D-FenderAJayMN, Any number of things, and none of those are ASTERISK.
06:22.06AJayMNok
06:22.20AJayMNfigured someone else may have had the same issue at some point
06:23.32[TK]D-FenderAJayMN, With no details of any kind.  Not likely.
06:23.44drmessanoAJayMN.. you're asking in the wrong place :)
06:24.42[TK]D-Fenderok, I'm off, later all
06:24.50drmessanolater
06:28.32Malkutlol
06:28.33Malkutthnx
06:30.59*** part/#asterisk AJayMN (n=mypocket@76.201.155.119)
06:37.28*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:39.14BBHoss~seen bkw
06:39.18jbotbkw <n=bkw@h-235-0.A189.cust.bahnhof.se> was last seen on IRC in channel #debian, 10h 21m 42s ago, saying: 'valdyn: apt-cache search apache2 ssl  suggest several packages. But which one installes the modules'.
06:49.05AndyGraybealanyone famillar with a ron paul phone call project on the web.. i swear it was either here or #freeswitch ... but they don't think so
06:49.13AndyGraybealor somewhere else.. another offshoot of asterisk
06:49.30BBHosscallweaver?
06:49.40AndyGraybeali don't think so
06:50.05J4k3~ron paul
06:50.06jbotZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT
06:51.33*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
06:51.34drmessanoMaharishi University of Management
06:51.43drmessanoI bet they don't teach telecom
06:51.49drmessanochan_om
06:54.26drmessanoRon Paul?
06:54.41drmessanoRon Paul phone call project... what is that?
06:55.25J4k300:50 < J4k3> ~ron paul
06:55.25J4k300:50 < jbot> ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT
06:55.40J4k3phone spam politics
06:55.44drmessanoOh
06:56.05drmessanoThat ZOMG line is mine, btw.. I have trademarks on it now
06:56.12drmessanoMatter of fact, consider yourself sued
06:56.22J4k3ron paul should pony up some cash and hire indians to do it
06:56.26drmessanolol
06:56.27drmessanoHAH
06:56.38drmessanoHALO, MY NAME IS.... JOE..
06:56.44J4k3I am from new york!
06:57.21drmessanoRon Paul has Digg.. who cares if they're all too young to vote
06:57.46J4k3wtf is digg
06:57.49J4k3is it like facebook?
06:57.55drmessanoOMG
06:57.58J4k3something else for fake-nerds to do?
06:58.00drmessanoDigg.com is ummm
06:58.09drmessanoIt's a "news" site
06:58.14J4k3ahh
06:58.16J4k3its one of those
06:58.18drmessanoIt's like an unmoderated Fark
06:58.20J4k3I hate the internet
06:58.21J4k3I just sell it
06:58.35J4k3I don't get high off my own supply
06:58.42drmessanoEverything is "BREAKING NEWS:  ZOMGGGGG"
06:58.56J4k3and then you sue them for ZOMG abuse?
06:59.29*** join/#asterisk steliosk (n=Stelios@85.75.198.88)
06:59.29drmessano"NEWS FLASH: APPLE TO ADD AN EXTRA 1GB TO MACAIR 2 !!!!ELEVEN!!!"
06:59.46drmessanoThats pretty close to a Digg headline
07:00.09J4k3haha damn
07:00.14J4k3thats some lame people
07:00.20drmessanoIt's hilarious
07:00.22J4k3but not suprising, the internet lacks :|
07:00.32drmessanoApple fanboys, ron paul fanboys, linux fanboys
07:00.48J4k3theres all these places with info, but nobody has any real info
07:01.09J4k3an internet-load of reporters and nobody out making news
07:01.10drmessano"BREAKING NEWS: UBUNTU TO ADD UPDATED VERSION OF NANO TO 9.09 BUSTY BEAVER!!!!!!!!ONES!!!1111!!!2!!"
07:01.24J4k3busty beaver... now *thats* my release.
07:01.41coppiceyou like 'em furry?
07:01.46drmessanolol
07:02.05J4k3beggers can't be choosers.
07:02.38*** join/#asterisk IPGHOST (i=IPGHOST@202.142.153.55)
07:02.41drmessanoI should start a Digg-like Asterisk news site
07:03.15J4k3ZOMG CHAN_SPOOGE GETS GISM CONNECTIVITY TO ASTERISQE
07:03.40drmessano"BREAKING NEWS:  AT&T ADMITS ASTERISK IS HURTING THEIR BUSINESS!!!!"
07:03.57sbingnerlol
07:03.59drmessanoLink to some article about AT&T embrasing VoIP
07:04.03J4k3BREAKING NEWS: Level 3 still sucks!
07:04.28drmessano"AVAYA LOSES MILLIONS DUE TO SUCCESS OF ASTERISK"
07:04.37J4k3AVAYA GOES AWAYA
07:04.45drmessanoNo, those arent lame enough... hmmm
07:05.01denonbreaking news: avaya is running asterisk on their new units
07:05.04drmessanoHa, got it..
07:05.13BBHossABE at that :)
07:05.16denon(and not releasing source)
07:05.24denon<rumors/>
07:06.02drmessano"1283 REASONS TO UPGRADE FROM ASTERISK 1.2 TO 1.^ (BEFORE IT'S TOO LATE!!!)!!!!"
07:06.09drmessano1.6*
07:06.27drmessanoThere has to be some drama
07:06.36J4k3BREAKING NEWS: UPGRADE TO ASTERISK 1.6 FOR Y10K COMPLIANCE
07:06.39denon1) we'll be taking the upgrade script off the servers tomorrow!
07:06.44drmessanolol
07:06.52Corydon76-digUh, all it takes for Avaya to lose millions (in sales) is for them to fail to get a single contract
07:06.55denonupgrade tonight or lose all your settings!
07:07.23Corydon76-digThat's less about other people coming on the market than it is about how massively expensive Avaya equipment is
07:07.38denonof course, small countries run their telcos on avaya..
07:08.00drmessano"BREAKING NEWS: IRAN TO START USING ASTERISK"
07:08.25denonDial(IAX2/iran/terrorist-queue)
07:08.28J4k3http://www.whoopis.com/howtos/telco-basics/beirut-phone-wiring.jpg
07:08.34coppicethe US army in Iraq is using Asterisk
07:08.36J4k3^^ awaiting asterisk upgrade
07:08.47J4k3also, need GPS locator for cat
07:08.54drmessanolol
07:09.11drmessanocat /pees/interrupts
07:09.28J4k3if I was a cat and saw that phone box
07:09.31J4k3I'd spray it, twice.
07:10.14drmessanoThe one thing you'll find about Digg is that Apple does NOTHING wrong
07:10.22denoncoppice: I set up some asterisk gear in iraq for soldiers to call their families at home
07:10.33drmessanoThe headline may not support it, but the comments will
07:10.39denonjust a simple bridge back to some some LD trunks we did for free on the US side
07:10.53denonon a guy's hardware over there
07:10.54drmessanodenon: that's awesome
07:11.00coppicedenon: they are doing more with it now, it seems
07:11.10J4k3nice, asterisk is good stuff.
07:11.12J4k3flexible, easy, etc.
07:11.16denonthat was a very long time ago, I dont think its in use anymore, 'cause those divisions came home
07:11.31denonI should disable it so nobody abuses the free LD heh
07:11.32J4k3hell, trixbox can be configured by the average shop mechanic.
07:11.41drmessanoahmadinejad loves Asterisk
07:11.44denonJ4k3: trixbox is really only good for a shop mechanic :)
07:11.49drmessanolol
07:11.53coppicedenon: well, someone is setting up something right now, needing MFC/R2 interconnect
07:12.06drmessanoMy cat loves Trixbox.. she can set one up in an hour
07:12.09denonah, yeah, doubt that's my rig heh
07:12.12J4k3denon: exactly, but worse than avaya?! :D
07:12.23*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)
07:12.25J4k3my cat knows what to use trixbox for
07:12.32*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)
07:12.37J4k3*scratch scratch scratch*
07:12.40denontrixbox works fine for some stuff .. just don't forget to change the litter
07:12.47drmessanolol
07:13.19drmessanoecho 127.0.0.1 *.fonality.com >> /etc/hosts
07:13.56denonyeah, the nice thing about trixbox is that it's a truly open system .. all your data, system details, usage info, etc .. totally open for fonality to do whatever they want with it
07:14.06drmessanolol
07:14.08drmessanoYep
07:14.22drmessano"Anonymous usage data"
07:14.34denonthose threads are entertaining
07:14.42denonat the beginning, they claim the processes don't exist
07:14.53denonthen they claim it's not personal data
07:15.03denonthen they claim it's optional, and well-documented
07:15.06drmessanoI was called a TROLL.. that wasn't entertaining to me :(
07:15.32denonJ4k3: two grids, 2 LP or diesel backup gens, ASCO transfers
07:15.32drmessano"We could have hidden it much better, but we decided not to"
07:15.36denonyeah
07:15.46drmessanoThat was my argument
07:15.46J4k3theres two ways I see to build it...  high capacity storage, or low capacity storage with redundant generation... leaning toward the second.
07:15.55J4k3denon: not possible here.
07:15.58drmessano"We could have hidden it much better, but we decided not to"   and "We didn't think anyone would be upset"
07:16.02denonJ4k3: you know, LP is *really* nice for long-term storage
07:16.10J4k3yeah, we've got LP onsite already
07:16.12denonengines burn clean, fuel doesn't get goofy
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07:16.19drmessanoIf they didnt think anyone would be upset, they never would have NEEDED to discuss hiding it
07:16.20J4k3I keep 100 gallons onhand at all times
07:16.23denonand it's easy to keep a week or more onsite
07:16.27denonadd a 0 :)
07:16.30J4k3what I love about LP on smaller engines is the fact you *can't* flood it
07:17.02denonnod
07:17.06J4k3I mean you can, but its not a wet flood, cranking it over a couple turns with the throttle open will clear it
07:17.12denonyou wouldn't know of such things, but it also runs pretty well in cold weather
07:17.21J4k3yeah
07:17.23denonthough our gen gear is indoors anyway
07:17.33J4k3can't always trust indoors to be warm.
07:17.45denonwell, we keep oil warm, as well as battery
07:17.57drmessanoWe've had a few problems with the tanks when they get below about 40%
07:18.20denondrmessano: yeah, some engines require a certain column inch volume ..
07:18.37denondepends on your tank size, type, etc
07:18.38drmessanoYeah.. It was a pain in the ass during the last few ice storms
07:18.47denonyou have an upright tank?
07:18.54drmessanono
07:19.08denonyour gen burn vapor or liquid?
07:19.28J4k3I'm thinking I could probably live with about 5kW of generation capacity
07:19.41drmessanoI'm not the expert.. but that was the issue.. I believe it burns vapor
07:19.52drmessanoand we were talking about putting something in line to change that
07:20.04J4k3the lowest temps we see here are -5F, should I worry?
07:20.04denonwell, vapor isn't bad ..
07:20.09denonno
07:20.23denonvapor's not bad, but you need a large enough tank
07:20.27drmessanoyes
07:20.35drmessanoI remember that
07:20.40sergeeany gprof experts here?
07:20.42denonit's not so much the size, but the amount of airspace
07:20.57drmessanoWe got two much larger tanks for next time
07:21.21drmessanoBiggest issue was keeping them topped off
07:21.25*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:21.27denon1,000 gal tanks are usually a good match for a couple decent gens to run a long time
07:21.49J4k3debating 12V or 24V system... 12V is nice due to the quantity of cheap alternators available
07:22.08denonJ4k3: why not just pump 480 or 240 or whatever into your mains?
07:22.14J4k3thinking of maybe a small 4 cylinder car engine set up on a sled..  I've helped build something similar before
07:22.19J4k3denon: cost, lack of need.
07:22.20denonsure, DC is more efficient, but..
07:22.28denonnice to have your workstation when the lights go out :)
07:22.31J4k3denon: well, it'll get inverted.
07:22.38J4k3my workstation is a laptop
07:22.48J4k3the only thing I miss is my desk lamp... and I plug it into the inverter now ;)
07:22.53denonhehe
07:23.01denonyou and Edison would get along well
07:23.18denonthat darn Westinghouse and his silly A/C nonsense
07:23.19BBHossanyone here using adhearsion?
07:23.23J4k3no
07:23.40J4k3I appreciate AC a lot :)
07:23.48J4k3DC scares the holy crap out of me, thats why I refuse to play with 48v
07:23.56J4k324v scares me pretty well
07:23.56denonhehe
07:24.09denoneh, why?
07:24.12J4k3I've been hit with everything 12v had to offer... it stung like hell but I suffered no damage
07:24.26J4k3electricution by DC is nasty
07:24.30J4k3you jam up and won't let go
07:24.32J4k3AC slings you out
07:24.44denonI'm pretty sure the electric chairs runs on AC ..
07:24.51J4k3yeah
07:24.55J4k3stepped up
07:25.02denoncourse you're a little constrained in that. .
07:25.13J4k3texas's first electric chair ran on a generic home pole pig wired backwards for step-up
07:25.32denondid you know that Edison was lobbying the govt to call that process "getting Westinghoused"?
07:25.54drmessanoBastard killed an elephant
07:25.55denon'cause he was trying to get the public to fear A/C
07:26.48J4k3but yeah, the human body can sink a lot of current/heat
07:26.51J4k3before it fails
07:26.57J4k3so yeah, the electric chair has to be pretty brutal
07:27.14J4k3and most electric chairs are pretty darned current limited
07:27.26J4k3they wanna make sure it looks good.
07:27.26denonThe Electric Chair: Like Cisco Call Manager, with a seatbelt!
07:27.27J4k3:|
07:27.30denonthere's a topic ..
07:27.37J4k3haha owned
07:27.56sergee"adterisk and electric chair" - friday 7/8 central on ABC
07:28.08sergees/adterisk/asterisk/
07:28.11drmessano"BREAKING NEWS: ZOMG, CISCO KILLS KITTENS!!!!!!"
07:28.16denonyeah, going from generators to electric chairs, perhaps we're a little off topic
07:28.31denonbut, cisco does kill kittens
07:28.35denonI read about it in a press release
07:28.39sergeeguys, please , enlight me
07:28.40J4k3denon: see, my idea is run it off an inverter
07:29.07drmessano"Cisco Call Manager is powered by the tiny hearts of slaughtered kittens :((((("
07:29.10sergeewhat is "central US time zone" - what is the offset from GMT
07:29.15J4k3sit a relay with time delay between the mains and my gear... basically let it fall onto the inverter immediately (standby UPS style) but delay (or wait for me to do it manually) switch back to mains
07:29.20denonsergee: -6
07:29.22denondepending on DST
07:29.42J4k3likely a 5k sustained (8-10k peak) inverter
07:29.58denonJ4k3: why not just toss an asco in?
07:30.13J4k3asco?
07:30.16drmessanoOnan FTW
07:30.20denonyeah, brand of transfer switch
07:30.21sergeedenon: thanks and one more question, how to read times 8/9, 7/8 ?
07:30.27J4k3onan costs a lot, but their stuff is nice
07:30.27denonit's UL listed
07:30.32J4k3more than I want to spend
07:30.34denonsergee: 9, 8 central .. the 9 is eastern
07:30.37drmessanoWe use all Onan.. good stuff
07:30.51denoner .. eastern? I dunno, some timezone
07:30.51J4k3denon: ahh...  a proper transfer switch would work... but the idea is there'd always be AC available from the inverter from battery
07:31.05J4k3as soon as theres an AC outage event, crank the motor and start generating DC
07:31.08sergeedenon: so 9/8c == 21:00 east coast, 20:00 central, am i right?
07:31.15denonsomethin like that yeah
07:31.16J4k3so you get nice clean 'electronically generated' AC
07:31.24J4k3instead of whatever a cheap-ish generator feels like generating
07:31.53denonJ4k3: well, I dunno, good gens usually have the engine and generator tuned pretty well to eachother
07:32.04denonI see a pretty consistent 59hz over a long time
07:32.11denon59.8-60 or something
07:32.33denonbiggest problem I see, is when people underpower their gen
07:32.36denonand the engine pulls hard
07:32.42denona poorly designed gen will just slow down
07:32.47denon(and obviously cycles go with it)
07:32.49J4k3well, I don't want to spend $5k on a $1k problem
07:32.50J4k3yeah
07:33.08denonsounds like you want lots of batteries from the telco
07:33.11J4k3the problem I've seen is when the generators themselves start failing...  good units catch it and unload... bad units destroy stuff
07:33.32J4k3thats what happened with my last generator...controller went nuts after about 3 hours, ate a few dozen psus.
07:33.33denonyeah
07:33.39denonug
07:33.46denonUPSs didn't catch?
07:34.09J4k3they tried
07:34.17denonon, off, on, off
07:34.19J4k3they didn't fail, oddly enough... they ate hardware behind them
07:34.23J4k3IDidn't trust them anymore, though
07:34.24J4k3yeah
07:34.35denongotta set the thresholds really tight
07:34.36denonsucks
07:34.47denongotta force em to just go hard to battery if things aren't perfect
07:34.55J4k3yeah
07:35.09J4k3but what I'm thinking is... if I run a 12V system off battery
07:35.15J4k3I can charge it off pretty much anything
07:35.28J4k3currently I use a car in case of long term outage.  it works.
07:35.33J4k3short term we just run off battery
07:35.53denonyou can rip the inverter/charger out of a UPS..
07:36.02J4k3yeah
07:36.13denonwith adequate cooling, I'm told those things will push a lot more than they rate em at
07:36.16J4k3I could get a decent sized UPS and throw a lot of battery behind it
07:36.30J4k3you can usually run them a lot longer than their rating
07:36.33*** join/#asterisk ik_5 (n=ik@85.64.203.142.dynamic.barak-online.net)
07:36.39J4k3a lot of cheap ones overheat due to lack of heatsink area
07:36.42denonif you cool em yeah
07:36.44J4k3if you over-battery them
07:36.51J4k3but yeah, if you use non-crap UPSes and fans, no problem
07:36.51denonand of course, make sure you vent your hydrogen ;)
07:37.03denonassuming you're probably going to use truck batts or something
07:37.10drmessanoThis sounds like a job for a honda generator and an APC power conditioner :)
07:37.42denondrmessano: or a symmetra, and a pallet of old batteries from his local scrap yard :)
07:37.44*** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
07:37.51drmessanoheh
07:37.57J4k3thats why god made tractor supply company.... for cheap plastic truck tool boxes (Battery boxes) and 2 gauge stranded heavy duty 'welding wire' by the foot
07:38.13denonhehehe
07:38.38J4k3you gotta go crazy with the wire when you're running inverters off 12V
07:39.02adeelwould an echo canceller running on an FXO port cause problems with faxing? and if so, would setting the faxdetection options in zapata help?
07:39.19denonadeel: most people disable echo can on native bridge fax stuff
07:40.00adeeldenon, alright, i'll try making sure that the echo cancel on bridged setting is properly set
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07:40.46adeelhmmm, it's already set to no
07:41.35denontake a look at the channel while it's up
07:41.44denonduring a fax or whatever
07:41.56adeelusing ztmonitor?
07:42.02denonnah, just in the console
07:42.25adeelwell the output seems to be normal, i can see the zap/8 channel connect to zap/g0 and the call being passed
07:42.32adeelbut for some reason, transmission fails
07:43.07denonare you looking at: core show channel zap/123-1 ?
07:43.31adeelwell, i'm running * 1.2 still on this box
07:43.43denonwell, sans the core
07:44.07*** join/#asterisk zeeesh (i=zeeesh@203.215.179.43)
07:44.31adeelwhich output should i be looking for...Echo Cancellation?
07:45.55denonit'll show if it's on or not during the call
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07:46.58denonyou'll see something like Echo Cancellation: 128 taps, Currently Off
07:47.23denonthat's realtime, during the call
07:47.29adeelyeah, i see that...but to make things more complicated, i'm using a commerical echo canceller (octasic softecho suite) because the built-in routines weren't cutting it
07:47.37denonIve gotta bail, see you guys later
07:47.42adeelthanks, see ya
07:47.49denonah, yeah, you'd need to check with them then, I dunno anything about it
07:47.58denonyou might also play with digium's HPEC
07:48.14denonit's free on cards currently under a year old or something
07:50.09denonhttp://www.digium.com/en/products/software/hpec.php
07:50.11denonanyway
07:50.12denonaudios
07:56.04ik_5hello, I have an asterisk box wich is a blackbox for me, that is I do not have access to the internals, it has few fxo and fxs, and I'm connected using ethernet (iax/sip), I wish to send a fax using the machine with the iax/sip throu the fxo, what components on the client side (linux btw) i require to have in order to achive the sending of a fax ?
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08:00.12JTik_5: it can't be done reliably like that
08:01.29ik_5JT, what does it mean ?
08:01.43DavisGrik_5 are U have t.38 support also ?
08:02.27ik_5DavisGr, no
08:03.07DavisGrpoor, but this is good aplication http://news.asteriskguru.com/2/9220/2008/1/9/Zoiper_softphone_now_supports_T.38_Faxing
08:03.38*** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
08:05.28DavisGrtry to find some fax over sip software / or buy some Linksys ATA fxs to ethernet then U can use your faxmachine but if U have large network then U will lose some fax's
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08:06.58ik_5thanks
08:09.16*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
08:13.08JTik_5: i mean it probably won't work
08:13.12JTespecially with analogue cards
08:14.37ik_5JT so i must have T1 support as well ?
08:14.53JTideally
08:15.00JTand faxing isn't that reliable in asterisk anyway
08:15.06JTand faxing isn't reliable over VoIP
08:15.14Frogzooanyone like to recommend a linux sip client?
08:15.32ik_5Frogzoo, twinkle
08:15.37Frogzooik_5: thanks
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08:24.59ik_5is there a way to use hylafax and iaxmodem to send the fax over iax to the fxo ?
08:25.56DavisGrit's depend how to make your system - if U will use hylafax then U can send fax over fxo usin hylafax print driver or using emails
08:27.07ik_5the probelm is, that i can't install anything on the asterisk with the fxo...
08:27.58ik_5so i must do things on the client side
08:28.22DavisGrIf U cant install anything then U can try to use Linksys ATA - but it fill work if U will not have large network from ATA to pbx
08:29.02DavisGrbut then U will need sip accaunt/extension for this ATA
08:29.31ik_5it's very small network, i can even make it almost direct connection ...
08:29.46DavisGrthen it will work
08:30.32DavisGrI have some cases where is one swich behind and its working
08:31.14DavisGrThe moust important is network load if U have Data & VoIP network the same
08:31.44DavisGrdont Use p2p software like torents dc++ etc.
08:32.13ik_5this is a dedecated network, so no one will use it :)
08:32.34DavisGrthen fine it will work!
08:32.57ik_5thanks, i'll look at the ata
08:35.38DavisGrLinksys SPA-1001 1xFXS ,  Linksys PAT2T - 2xFXS but for them U will need use the fax machine ofcourse!
08:35.47JTDavisGr: is it that hard to type "you"? ;)
08:36.05J4k3joo
08:36.29JTalso, saying it WILL work is a big too optimistic
08:36.34JTit "might" work
08:37.33DavisGrJT  bouth end of last year clients from one voip commpany there was such cases and its work
08:38.13DavisGrFor faxs I prefer callweaver becouse he support T.38
08:38.42JTyes, but you definitely can't guarantee it will work for him, especially if he;s using analogue digium cards
08:40.07DavisGrwhere is problem width analog digium cards - there in market is also copies of digium and they also work!
08:41.34DavisGrJT for 100% I cant guarantee but I can say I had solutions where in that way working!
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08:49.05JTDavisGr: digium analogue cards are not good at faxing
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08:52.12scooby2Whats the best way to debug why zaptel or the te412p module is kernel panic'ing centos? mentions smp.c
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08:58.14DavisGrJT what is the best card for analog lines
08:58.42DavisGrscooby2 witch kernel you have
08:58.56JTDavisGr: a real fax machine or fax modem
09:00.00DavisGrOK :) but if you need handle inbound and outbound calls via asterisk
09:00.17DavisGrwith the same line
09:00.20JTdo what sensible people do
09:00.33JTand avoid it unless you have digital or a high end card
09:01.27J4k3efax!
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09:02.00DavisGrwi JT
09:02.45JT?
09:03.33J4k3vi JT
09:03.47ik_5Jt, so maybe you'll have a different solution for the problem, I have to update 30 people with information regarding things they bought, or things they requested to know, and they only use faxes... how would you computerize this without having a modemfax ?
09:03.47*** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk)
09:05.37JTik_5: buying an analogue line is probably the easiest
09:06.08ik_5JT, at this time I'm trying to use only things (in hardwre) I already have
09:06.17DavisGrJT it's depend of situation where are you!
09:07.44DavisGr<PROTECTED>
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09:08.34ik_5DavisGr, it's a flash based pbx so any change can destroy things...
09:09.00DavisGrto bad
09:09.39ik_5yes, the problem is that I do not have a lot of control over things... other wise it would have been less chanlnging and "too easy" ;)
09:14.15*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
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09:16.03Frogzoodoes fax on analogue digium always have problems?
09:16.22JTmainly on the single and quad port card
09:16.47*** join/#asterisk the_5th_wheel (n=edd@64.251.30.240)
09:17.01FrogzooJT: hardware software or driver issues?
09:17.17the_5th_wheelHi. Can anyone reccommend me a billing system, wich just log wich calls are made from wich sip user?
09:18.21*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)
09:18.24scooby2DavisGr: centos 5.1 - 2.6.18-53.1.6.el5
09:21.45FlatFootmorning all
09:21.47DavisGrscooby2  I was the problem last summer width enabled smp on slackware width 2.6.16-20 kernels when smp switching off then things start work but on latest kernel versions I havent problems
09:21.59*** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-2c66359af2644e4e)
09:22.21FlatFootcan anyone help with iax2 one way breakdown probs ? http://www.pastebin.ca/894824 shows a bit of the debug
09:23.00FlatFooti have got jitterbuffer turned on and it is a trunk
09:23.42FlatFootboth ends are running 1.4.17
09:25.29JTFrogzoo: hardware
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09:27.51jeremy_goej, oej!!
09:27.54FrogzooJT: ugh, that kinda sux
09:28.17oejjeremy_g: Jeremy, jeremy!
09:28.22jeremy_g:)
09:28.47jeremy_gI have to put asterisk in front of this IMS (CSCF) with a big operator
09:29.21jeremy_gI am making a feasibility if asterisk would be usable in this case. I am in sweden ;) as you know.
09:29.42Sajjad_Ali_Mushtanyone knows "how to capture the QoS parameters from SIP, ZAP and CAPI channels for completed calls"
09:31.20jeremy_goej: Would the IMS specific headers like P-Asserted Identifity, Visited Network Id, "reg event" confuse asterisk?
09:32.34scooby2DavisGr: thx
09:33.01scooby2can anyone point me to a 1.4 IVR example? trying to upgrade from 1.2 and it is not going well.
09:33.32ik_5DavisGr, JT, thank you for the help, bye
09:34.24oejjeremy-g: Not confuse, just be ignored
09:40.21synthetiqwhere can i find the text2wave application?
09:40.34*** join/#asterisk _gm (n=gmustafa@58.27.175.222)
09:46.43mvanbaaksynthetiq: festival
09:50.59*** join/#asterisk matlads (n=matlads@217.113.73.39)
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09:52.23yangI wonder has anyone been using munin plugins for graphing asterisk processes?
09:54.23matladshello
09:54.31Frogzooyang: cacti is nicer than munin or mrtg
09:54.39Frogzooimo
09:54.44matladsone question, can I do a SET VARIABLE in a hangup script?
09:55.54yangno, cacti is a pain to setup
09:56.11yangsnmp!"#$WR
09:56.55XnOSXhave any probs with asterisk and kernel 2.6.24-1-686?
09:57.29jeremy_goej:it would ignore them even if these are additional headers within a proper sip message?
09:58.24jeremy_goej:you have read the TS 24.229 right? You know that these are just really sip headers within the regular SIP methods like INVITE, BYE etc.
09:58.44jeremy_goej:so would an additional header like above cause the * to ignore the whole properly formatted SIP?
09:59.34jeremy_gyang:cacti is not a pain to setup, we run it here and its a breeze
09:59.41jeremy_gFrogzoo:yup
10:04.37jeremy_goej:reply me
10:04.51oejWas offline...
10:05.15oejNo, we just ignore the headers
10:05.21oejI can't remember TS 24.229 - URL?
10:05.36jeremy_ghttp://www.3gpp.org/ftp/Specs/html-info/24229.htm
10:08.07*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
10:08.07*** join/#asterisk disposable (n=asdf@87-194-169-5.bethere.co.uk)
10:08.49oejjeremy_g: If you need any work done to make asterisk more compliant with this, you know where to find me :-)
10:09.34nixguyany nice webgui tools for monitoring asterisk calls n stuff with?
10:12.13jeremy_goej:gimme ur contact and i dont even know if there is a need for more work to be done.
10:13.36jeremy_goej:my understanding is that * wud still process the invites coming in from cscf no matter what additional headers are there, both wud happily talk sip as long as cscf does not expect * to talk those additional headers.
10:14.38tzafrir_homeXnOSX, you'll probably need zaptel from svn at this point
10:14.42mvanbaakjeremy_g: you can add extra headers from the dialplan
10:18.10*** join/#asterisk ToTo (n=ToTo@207.176.6.66)
10:19.56jeremy_gmvanbaak:yup i know
10:21.58oejjeremy_g Yes, that's right
10:23.39XnOSXhey friends! what is the best kernel for asterisk
10:23.40XnOSX?
10:23.48XnOSX2.6.18?
10:24.09jeremy_g:D
10:24.19jeremy_gCol. Alfred Douglas
10:27.11mvanbaakXnOSX: any 2.6 that you are comfortable with
10:29.05*** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au)
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10:32.28L|NUXhello
10:32.47L|NUXcan some one help me with MoH mms stream ?
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11:07.13sergeyHi. Is callcentr on * can "speak" number into line and/or how mach time till answers by op ?
11:07.19*** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar)
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11:22.01Sweeperanyone know how i can see some logs on the aastra 9133i?
11:23.24Sweeperhaving some issues getting it to register, and don't have access to the sip server
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11:41.44synthetiqwhere can i find the text2wave application?
11:42.23synthetiqbecause app_festival is built but i cant find the script or binary for text2wave
11:52.46*** join/#asterisk tnt_ (n=tnt_@8.253-244-81.adsl-static.isp.belgacom.be)
11:53.26*** join/#asterisk ArchSSM (n=tommy@host-81-191-139-130.bluecom.no)
11:53.28tnt_Hi everyone. I have an issue with a B410p. Just to be sure, the leds must be green even without asterisk launched, just mISDN is enough to sync right ?
11:53.53ArchSSMtnt_: That is correct.
11:54.11tnt_ArchSSM: And if it's red ... what could that be ?
11:54.36ArchSSMthen it's out of sync. probably a driver issue
11:56.41*** join/#asterisk nighty^ (n=nighty@p3125-adsau17honb13-acca.tokyo.ocn.ne.jp)
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12:08.13phixhi
12:08.16phixlets all help me
12:08.21phixnow please k thnx
12:08.25phixor whaen ever
12:08.29phixprefererably npow
12:08.31phixthat would be great
12:08.44ArchSSMIf you have a question, just ask.
12:12.14tzafrirjbot, tell phix about ask
12:17.03*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
12:20.07synthetiqwhere can i find the text2wave application?
12:20.30synthetiqbecause it is not built by default..
12:21.04uwehello, im having problems with g729 sound quality ... ive setup a client to use g729 ... the recipient hears the voice very well, but the caller hears the recipient with very bad voice quality ! and i have no idea how to debug or tune g729 ... if it possible at all, another thing, i suppose translating from g729 to other codecs goes always through slin , so it doesnt matter if i translate from gsm to g729, or is there a better combination ?
12:22.18uweits g729 <=> asterisk <=>gsm<=>landline .... landline hears g729 well, but g729 doesnt hear landline well
12:26.25J4k3I do not believe g729 and gsm are complimentary
12:26.31J4k3I think the end result is very awful audio
12:27.31J4k3they'd be better off running asterisk on their end transcoding from gsm to g711, if their equipment doesn't natively support gsm and doesn't have the bandwidth for g711
12:27.54J4k3(I'm sure it supports g711 if it supports g729)
12:30.19SteveTotaroanyone know bristuff that can help?
12:30.24*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
12:30.25SteveTotarowith qozap?
12:31.04ArchSSMSteveTotaro: Ask, and we'll answer as best as possible
12:31.16uweJ4k3, ... sorry, i dont understand you very well, i can change the gsm encoding to something else, but not g729
12:31.35uwewhat do you suggest to set it to ... or do you mean it doesnt make a diffrence
12:31.47tnt_ArchSSM: And what kind of driver issue could do that ? Try another version or ?
12:32.39ArchSSMtnt_: That is very hard to say. Do a pastie on a pastiesite on the appropriate logs, and comment on what driver you're using
12:35.20tnt_ArchSSM: http://pastebin.com/m65ff81df
12:35.34tnt_And I'm using mISDN-1_1_7_2
12:35.50ArchSSMwhat card was it?
12:36.04JTuwe: compressed codec, to another compressed codec, don't be surprised if the end result is not very good, is what he's saying
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12:37.25tnt_ArchSSM: b410p
12:37.34tnt_used in Belgium ... (if that matters :)
12:38.29*** part/#asterisk Sweeper (i=sweeper@66.221.78.1)
12:38.44ArchSSMshouldnt :)
12:39.24ArchSSMwell... at least try to turn on debug and se if you get anymore information
12:40.05tnt_What value to use for debug ? just 1 ?
12:41.38ArchSSMyep
12:44.26uweJT, what codecs are not compressed, what is the raw format ? ulaw and alaw ?
12:44.39uweg711 ?
12:46.09RoyKeverything's compressed somehow
12:46.30RoyKg.711a/u is 13/14bit 'compressed' to 8bit
12:46.33uwehmm ...
12:47.00RoyKbut g.711a is the closest to 'raw' (i think)
12:47.06*** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv)
12:47.10aiksa[LV]howdy everyone
12:47.33aiksa[LV]anyone comming to asterisk con in NL?
12:47.50tnt_ArchSSM: http://pastebin.com/m2cfd0f68 That doesn't really speak to me ... :)
12:48.49ArchSSMSame here really
12:49.43aiksa[LV]I had another question in mind, is it possible to automatically move to the next dialplan priority a caller who had been waiting in queue for a given amount of time?
12:51.05*** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it)
12:51.27*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)
12:56.01mostyaiksa[LV], i believe there's a queue option to specify that
13:00.09Frogzoousual pbx's allow you to dial zero to get an 'outside line' - how best to do this in the dialplan?
13:00.23*** join/#asterisk Spyder12345 (n=bob@169.139.217.48)
13:01.12mostyFrogzoo, it's simpler to just use local extensions which aren't confused with regular telephone numbers, like 3 digit local extensions. then you don't need any of that crap
13:01.12cpminteresting, all the pbxs I've worked with are dial-9
13:01.23Spyder12345anyone in here know anything about dialing sip via tcp in 1.6?
13:02.51Frogzoothanks mosty, that helps
13:03.10*** join/#asterisk shinao1 (n=shinao1@41.221.165.57)
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13:05.26nebojsajsimichi all
13:05.52nebojsajsimic<PROTECTED>
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13:11.41mostyis there an asterisk management interface event for failed authentication of sip clients?
13:12.10nebojsajsimicno
13:12.20nebojsajsimicit normal work from exten
13:12.27*** join/#asterisk lirakis (i=lirakis@66.252.24.133)
13:12.45mostynebojsajsimic, huh?
13:14.18nebojsajsimici dont use autentication all is blank
13:14.24nebojsajsimicit is local system
13:14.57*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
13:15.38mostynebojsajsimic, i wasn't answering your question about AGI, i was asking about AMI
13:16.09*** join/#asterisk MikeBest (n=as@92.10.98.110)
13:16.29MikeBesthello
13:16.57nebojsajsimichi
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13:17.26MikeBestanyone experinced in asterisk?
13:17.55nebojsajsimicmosty: i have configured GSM GW in sip
13:18.05ArchSSMMikeBest: Probably quite a few here. Just ask if you have a question
13:18.08nebojsajsimicsip.conf
13:18.22MikeBesti am a total newbie
13:18.28nebojsajsimicme tooo
13:18.31MikeBesteven asking questions is difficult
13:18.33nebojsajsimicbut i try-ing
13:18.35nebojsajsimic:_
13:18.44ArchSSMMikeBest: In that case, you should read up a bit first.
13:18.48MikeBesti just have a project in my mind
13:18.52aiksa[LV]mosty, the only cooresponding option seems to be servicelevel, but I am not sure if it will disconnect user fro queue
13:18.58MikeBestand looking for some advice
13:19.13aiksa[LV]thats seems like an option for statistics only
13:19.24ArchSSMMikeBest: Sure.. try us
13:19.41mostyaiksa[LV], i want sip client authentication errors
13:20.01mostynebojsajsimic, what are you trying to do with agi?
13:20.13nebojsajsimicjust to make a call
13:20.16aiksa[LV]mosty: - i was referring to that queue issue
13:20.51nebojsajsimicDial(SIP/number@GW,r,15)
13:21.11aiksa[LV]anyone else, has any ideas? I really would not like to keep those timers in some kind of °service connected through AMI
13:21.24mostyaiksa[LV], there is a timeout option to the queue command, have you tried that?
13:21.27*** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254)
13:22.13aiksa[LV]mosty: nope, its not what i want
13:22.30aiksa[LV]timeout looks at how long agents phone have rung
13:22.31atis_worknebojsajsimic: you should also not write("Dial") directly, but give AGI commad "EXEC Dial..."
13:22.45mostyaiksa[LV], what are you trying to do?
13:22.45*** join/#asterisk angryuser (i=nononon@df01t2-212-194-235-109.d4.club-internet.fr)
13:22.58aiksa[LV]ok, here comes explanation:
13:23.42atis_worknebojsajsimic: please check http://www.voip-info.org/wiki/view/exec
13:23.43tuxfoohttp://www.voip-info.org/wiki/view/Asterisk+monitoring
13:23.58nebojsajsimicThanks!!!!
13:24.06tuxfoosee if that gets you what you want or gets you started in the right direction
13:24.15atis_worknebojsajsimic: i think also Dial options should be separated with space from Dial
13:24.17aiksa[LV]if a user joined the queue, but for lets say 5 min. he hadnt reach any of the agents (too long could be one of reasons) - he gots thrown out of the queue and proceeds to next priority
13:25.52nebojsajsimicFeb  7 14:24:06 WARNING[13425]: res_agi.c:1115 handle_exec: Could not find application  i get this
13:25.56aiksa[LV]mosty: that was the funcionality I wanted to achieve
13:26.31mostyaiksa[LV], the timeout in the Queue command does that, doesn't it?
13:27.26aiksa[LV]I guess - not, it counts seconds while a user had an interaction with agenty, but agent never picked up
13:27.35lirakismorning
13:27.42*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:27.42*** mode/#asterisk [+o lmadsen] by ChanServ
13:27.42nebojsajsimicmorning
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13:28.06*** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net)
13:28.27mostyaiksa[LV], "'timeout' sets the time in seconds that a call will wait in the queue before it is routed to the next priority in the dialplan" - isn't that what you want?
13:30.15aiksa[LV]wow exactly
13:30.15*** join/#asterisk LakeSolon (n=blake@63.231.182.86)
13:30.39aiksa[LV]mosty: but asterisk book SE gives different explanation for timeout parameter in queues.conf
13:30.49aiksa[LV]or this is not a queues.conf parameter?
13:30.55*** join/#asterisk MaartenB (n=Maarten@h8441243087.dsl.speedlinq.nl)
13:30.59MaartenBhello everyone
13:31.05mostyaiksa[LV], i'm talking about an option the the Queue application, not the setting in queues.conf
13:31.34MaartenBI would like to have a queue systems but without the trouble of agents logging on and off, I just want to have phones available for pickup by using the DND button on my phone, is that possible?
13:31.48*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)
13:32.03aiksa[LV]mosty: Oh, i see it now. Stupid - me. I was thinking you referred to a config file.
13:32.06aiksa[LV]sorry
13:32.21mostyaiksa[LV], i did originally
13:32.46atis_worknebojsajsimic: try issuing "agi debug" in CLI to see what you're sending
13:33.11aiksa[LV]MarteenB - can you modify what DND button does?
13:33.13*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)
13:33.14ZaVoidmorning
13:33.25aiksa[LV]what extension to ring upon press?
13:33.48aiksa[LV]mosty - and stupid me, never to look at aparemeters which Queue() takes .. :P
13:34.11hmmhesaysit is morning
13:34.13lmadsenmorning
13:34.38hmmhesaysI'm about to crack my cell phone open and modify the battery case to use AAA batteries
13:35.04cpmjust kludge up a power adapter, it's easier
13:35.11cpmand you can use D cells
13:35.44hmmhesaysI can make triple aaa's look fancy and the mah rating is still 3x what the original battery is
13:36.56hmmhesaysand I don't have an extra power supply to cut up
13:37.10hmmhesaysAND the e815 psu connector sucks big time
13:37.15eric2anyone ever use attractel's t.38 faxing software for use with asterisk?
13:37.19cpmyeah, they do suck
13:37.55hmmhesaysI have a month left before I get a new phone, I this one eats batteries
13:38.12mvanbaakugh, I HATE it when I cant get it working
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13:39.32yangI would like to enable parked call the *8 string takes the call...I have these values, but it doesnt work well yet - http://openpaste.org/en/5008/
13:39.59lmadsenanyone know if the D() flag to Dial() accepts 'w' as a wait pattern?
13:40.06coppiceeric2: a few people seem to have tried it
13:40.25hmmhesaysyang what are you asking?
13:40.25eric2coppice: is it the way to go?
13:40.37lmadsenyang: did you enable tT (pls check to see what they mean with 'core show application dial')?
13:40.40hmmhesaysI could use a service manual fo rthis phone
13:40.54hmmhesayslmadsen,  kK is the 1 touch parking flag
13:40.56yanghttp://openpaste.org/en/5009/
13:41.09coppiceeric2: well, the people I know who tried it abandoned it. I don't know their reasons, though
13:41.13lmadsenhmmhesays: ahhh right... stupid Justin for waking me up early
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13:41.18eric2interesting
13:41.23*** join/#asterisk x86 (n=x86@p3m/member/x86)
13:41.37eric2would be good to find out the reasons for abandonment
13:42.11coppicethey might have just been they were too tight to pay at the end of the evaluation period, or it might be troublesome. I have no clear idea
13:42.13yanglmadsen: i need to attach tT to each DIal string at the end
13:42.43tuxfoodoes anyone know how to get the wildcard masks to work on an inbound zap line?  I can get an exact match to work, but the wildcard masks don't seem to work.  I want to intercept 800 numbers - I have s/8XXXXXXXXXX as my mask but it ignors it. - thanks
13:42.57lmadsenyang: i was wrong.. it's 'kK'
13:42.58lmadsen<PROTECTED>
13:42.58lmadsen<PROTECTED>
13:43.11coppiceeric2: it does appear they offer evaluations, though, so you could try it
13:43.20mostytuxfoo, patterns start with _
13:43.41eric2price is a tad expensive, the euro is high
13:43.55tuxfooI tried that as well and it did not work  s/_8XXXXXXXXXX
13:44.06coppicei've never actually seen a price for it
13:44.06mostytuxfoo, what is the s/ in there?
13:44.09eric2looks like they offer their licensing on a per channel basis
13:44.10lmadsenI paid 19 pounds for goosync and it's well worth it :)
13:44.19yanglmadsen: I have rt enabled (no T) http://openpaste.org/en/5010/
13:44.28yanglmadsen: ah ok !
13:44.33yangadding k there
13:44.56coppiceeric2: I do the same with my T.38 software - $0 per channel :-)
13:45.10eric2what's your setup like?
13:45.16lmadsencoppice: you should really up the price
13:45.33eric2are you even using t.38?
13:45.51coppicelmadsen: you mean like a 10% increase, or something
13:45.59tuxfoos/1234567890 works as an exact match, but the wild cards don't.  I really don't want to add every number,so I thought I could just mask it.  But it is not working that way
13:46.04eric210% of 0 is still a bad number   :)
13:46.05lmadsencoppice: heck... 200%!
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13:46.35lmadsenshhhh.... stop talking about him
13:46.52eric2oh great, another cannuk
13:47.03mostytuxfoo, are you trying to match s and 1234567890 extensions in that example?
13:47.19lmadsenmosty: no, he is trying to patch a CID with that
13:47.54yanglmadsen: http://openpaste.org/en/5011/
13:47.56lmadsentuxfoo: s/_8XXNXXXXXX
13:48.15lmadsentuxfoo: although I haven't actually tried to see if it works
13:48.54mostytuxfoo, you could use GotoIf and a regex expression if that doesn't work
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13:49.23hmmhesaysthere seems to be an extra terminal on this e815 battery
13:49.36hmmhesaysI have ground, <something>, +, +
13:50.44hmmhesaysI guess a trip to radioshack is in order
13:51.35tuxfoos/ is my start so certain number get a menu, while others just go striaght through.  That part is working. For, example, s/123456789 is my cell phone.  When it calls in to my system I get a menu to administer the system, while a unmasked nmber just enters the system and rings my wife's phone.  I would like to take it a step further and mask numbers to other extensions or ZAPTELLER them without having to know every 800 number.
13:52.03lmadsentuxfoo: did you try the pattern match in the CID match or not?
13:52.16yanglmadsen: its not working (quite) yet
13:52.31*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:52.50tuxfoono I did not try that since an exact match seems to work fine,  I just assumed I could pattern match.  I will try that
13:52.59yangi get nothing to pick up error
13:55.26yangDo I have to apply http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups  to be able to pickup the phone with *8 ?
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14:01.11mostymodules.conf
14:01.32mostydoes "show features" in the console show anything?
14:01.32*** join/#asterisk beek (n=klinebl@65.211.106.243)
14:03.07yangmosty: no modules called feature in /usr/lib/asterisk/modules/
14:03.30mostyyang, no res_features.so?
14:03.51*** join/#asterisk AndyGraybeal (n=andy@node50.34.251.72.1dial.com)
14:04.10yangmosty: http://openpaste.org/en/5012/
14:06.59yangmosty: and here are my extensions.conf http://openpaste.org/en/5010/
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14:11.41plikyang: features.conf only gets loaded at start - not reload
14:12.31b1ch0<PROTECTED>
14:12.56[TK]D-Fenderb1ch0: "core show function CHANNEL"
14:13.51plikgood morning [TK]D-Fender
14:14.07[TK]D-Fendermornin'
14:14.19yanghi TK !
14:14.40yangplik: yeah i tried with restart too
14:15.31[TK]D-Fenderyang: [2008-02-07 14:47:04] NOTICE[3252]: chan_sip.c:13957 handle_request_invite: Nothing to pick up for 4dcf5b624542e8e3@10.105.2.64 <-- I'm betting you didn't even set your call-groups.
14:16.04*** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com)
14:16.18[TK]D-Fenderyang: Go show us your sip.conf for those 2 phones.
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14:18.08yang[TK]D-Fender: yeah i havent , however there arent any Callgroups in features.conf, like i saw on http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
14:19.20[TK]D-Fenderyang: thats because those are the VALUES you have to set in SIP>CONF <--
14:19.38[TK]D-Fenderyang: they will not appear in features.conf, but they are USED by it
14:20.22minteeis there a way to write DB functions in the asterisk console?
14:21.09[TK]D-Fendermintee: which DB, and what do you mean by "write functions"?
14:22.32yang[TK]D-Fender: http://openpaste.org/en/5013/
14:22.50synthetiqfor asterisk agi, the only application that works is exec and set_extension, but nothing else works, get_variable, noop, etc .... anyone have a clue why?
14:22.57yanghttp://openpaste.org/en/5012/ & extensions.conf http://openpaste.org/en/5010/
14:23.16yang[TK]D-Fender: I just printed Asterisk book from page 50-80
14:23.20_gmsynthetiq: can you show your code
14:24.04[TK]D-Fenderyang: You did not set callgroups & pickupgroups for ANY of your phones.
14:24.17[TK]D-Fenderyang: those values you saw belong in your peer entries
14:24.54[TK]D-Fenderyang: otherwise they won't know which calls theyr even permitted to "pick up".  Otherwise you could have people randomly hijacking other peoples calls.
14:25.32synthetiqmy $digit = $AGI->wait_for_digit(60); $AGI->set_extension($digit);
14:25.40synthetiqplain and simple
14:25.40yang[TK]D-Fender: so I must apply Callgroup=1 to every sip entry
14:25.54[TK]D-Fenderyang: BOTH.
14:26.01yangok
14:26.20yangfor the pickupgroup i assume i need to add 60-80
14:26.24yanginto each sip
14:26.35synthetiqor i try to do noop on variable using : $AGI->exec('read','TARGET_USER||1|||60'); $AGI->noop('\${TARGET_USER}');
14:27.02synthetiqnasicly i want to do processing on a dtmf digit
14:27.06synthetiqbasicly
14:27.11synthetiqim using Asterisk::AGI
14:27.14[TK]D-Fenderyang: no.
14:27.33*** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net)
14:27.45synthetiqi also tryied the STDIN method of WAIT FOR DIGIT  and no luck
14:27.52[TK]D-Fenderyang: go read the page on those 2 values again, you are failing to understand the basics of this.
14:28.09synthetiqactually the script exits when waiting for a dtmf digit.....
14:28.23synthetiqso the only thing that works is $AGI->exec('read','TARGET_USER||1|||60');
14:30.11yang[TK]D-Fender: each sip needs a different Callgroup like Callgroup 1 for ext. 60 and Callgroup 2 for ext. 61, but pickupgroup is then 1-20 for all SIPS
14:30.17yangas i understood?
14:37.06yangok seems i got it right now:)
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14:39.05yang[TK]D-Fender: do you happen to know the "trick" about Grandstream BLF lights ? I have enabled the Subscribecontext=BLF and I can see the green lights and i can also dial those extensions by pressing the lights, but i dont see the occupied light (red)
14:39.43mostyyang, do you have sip hints in your dialplan?
14:40.36yangsure i do
14:40.44yangthey are in BLF context
14:41.35yangexten => 60,hint,SIP/60 ; Jozi
14:41.35yangexten => 61,hint,SIP/61 ; Fax
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14:43.39yangWe ran out of toner right in the middle of printing the Asterisk manual :/
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14:47.05mintee[TK]D-Fender, sorry, I had to walk away.  I mean I just wanna query some things from the asterisk DB from the cli, and not have to write it in an extension for testing...
14:47.07[TK]D-Fenderyang: I seriously doubt you need more that 3-4 pickupgroups/callgroups in your system
14:47.22[TK]D-Fendermintee: "help database"
14:47.44minteeoh, thanks
14:47.55*** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose)
14:48.32drakohaving bad times with ooh323
14:48.38*** join/#asterisk funxion (n=x@63.214.236.169)
14:48.48[TK]D-Fenderyang: BLF could be an incorrectly configured phone.
14:49.18yang[TK]D-Fender: it works with a vlines version of asterisk...when we plug the phones there, and it doesnt work with my asterisk configuration
14:49.32*** join/#asterisk freezey (n=freezey@gw.mypublisher.com)
14:49.53drakohttp://pastie.caboo.se/148665
14:50.19[TK]D-Fenderyang: I'd have to see the whole mess, but please try to tackle 1 problem at a time.
14:50.44yang[TK]D-Fender: ok , next week then
14:52.10*** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp)
14:52.28yangthanks !
14:53.09*** join/#asterisk unenough (n=jfds@CBL217-132-95-18.bb.netvision.net.il)
14:54.03unenoughhow hard is it to implement a module that interacts with audio in real time? meaning it should receive the decoded audio stream and be able to write to it as well
14:54.27zeeeshmissing module will anybody guide the name "agi.pm" "Can't locate Asterisk/AGI.pm in @INC (@INC contains: /"???????
14:54.31*** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp)
14:54.48mostyzeeesh, did you install that perl module?
14:55.58[TK]D-Fenderoh wow, awesome breaking news!!!! 2008-02-07 - AsteriskWin32 0.66 released build from asterisk 1.2.26.2, with X100P support.
14:56.13drakoHad to drop call because I couldn't make SIP/101-081c6590 compatible with OOH323/
14:56.27[TK]D-FenderNow Windows shmucks can use WinModems and asstricks!
14:56.31zeeeshi hv installed more than 15 different perl modules ... but i am unable to find by the name of "AGI.PM" so from where can i get
14:57.08[TK]D-Fenderunenough: Go look at an * app that does this already and see
14:57.59unenough[TK]D-Fender, if I knew which one to look at...
14:58.37[TK]D-Fenderunenough: Pick any that listen, and any that play back.  Look for the voice changer patch as well (GOOGLE)
14:58.44mostyzeeesh, try #perl since that problem really has nothing to do with asterisk
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14:58.56the_5th_wheeldoes anyone kbnow if there are something like a ATA avaliable, but with isdn lines?
15:00.13[TK]D-Fenderthe_5th_wheel: I've heard of BRI channel banks once or twice, but this idea is virtually unmentioned in here.
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15:04.05the_5th_wheelWhat are my options then to connect an analouge PBX to an asterisk pbx?
15:04.35[TK]D-Fenderthe_5th_wheel: Anything else clearly.  VoIP, PRI, Analog.
15:05.20mostythe_5th_wheel, what kind of interfaces does the old pbx have?
15:05.48drakothe_5th_wheel, b410p ?
15:05.53drakotdm b410p
15:06.12the_5th_wheelIm trying to connect various of our offices to the asterisk system. It varies, some have analuge trunks, some have bri trunks, some have PRI trunks avaliable
15:06.22[TK]D-FenderAh, you could use a BRI card for interfacing with *, but you mentioned ATA.
15:06.46mostythe_5th_wheel, you can get pci/pci express PRI, BRI and TDM cards
15:06.53[TK]D-Fenderthe_5th_wheel: For which there are a number of vendors
15:07.13the_5th_wheelSo i can connect a PRI card directly to the bri trunk of a pbx?
15:07.31mostyno
15:07.39[TK]D-Fenderthe_BRI to BRI
15:09.11the_5th_wheelIm not following.
15:09.21*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:09.25the_5th_wheelHow is that going to help me connect their PBX to the asterisk server?
15:09.57drakoany idea, [Feb  7 10:05:30] WARNING[3234]: app_dial.c:1685 dial_exec_full: Had to drop call because I couldn't make SIP/101-081b7bc8 compatible with OOH323/89.149.164.34-3de3
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15:10.18[TK]D-Fenderthe_5th_wheel: If they have BRI port, you can put a BRI card into your * server and connect them.
15:10.34minteedatabase put\get at the cli doesn't deal with the mysql database does it?
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15:10.57the_5th_wheel[TK]D-Fender: then i would need to go over telecom infrastructure
15:10.58[TK]D-Fenderdrako: Here's an idea, show us the entire call attempt with SIP and H323 debug so we can actually see the CAUSE, not just the warning.
15:11.04[TK]D-Fenderdrako: PASTEBIN is your friend
15:11.10the_5th_wheel[TK]D-Fender: the idea is to route things over the internet
15:11.12mostymintee, no. try func_odbc
15:11.40[TK]D-Fenderthe_5th_wheel: You are jsut not getting it.  You need to connect the 2 BOXES together.  Telling the PBX to send calls to * to deal with is another matter.
15:11.59minteemosty, that's not the problem really, i'm just curious as to what kinda DB it's dealing with them.
15:12.16[TK]D-Fenderthe_5th_wheel: they need to TALK first.  then what you can tell your PBX to do to send calls over the connection you now have with * is another matter.
15:12.20mostymintee, berkeley db, version 1 or something i think
15:12.30[TK]D-Fendermintee: No, it does not
15:12.49[TK]D-Fendermintee: those commands are for AstDB only.
15:13.15[TK]D-Fendermintee: If you want to manipulate MySQL, use "mysql" at *nix DCLI like the rest of the universe
15:14.09the_5th_wheelaso you are saying i put a PC into the small office, with one of these BRI cards, and have the PC route the calls to asterisk, but how do i connect to the standard PBX, what do i conect it to?
15:15.14mostythe_5th_wheel, an asterisk box with a BRI card can connect to another pbx that has BRI ports. i don't understand what else you are asking
15:15.14[TK]D-Fenderthe_5th_wheel: what is this EXTRA  PC you're talking about?
15:15.47[TK]D-Fenderthe_5th_wheel: you put the card into your ASTERISK server.  You plug the ports on this card into your existing PBX.  Whats so hard to understand here?
15:15.59[TK]D-Fenderthe_5th_wheel: there is no second PC.
15:16.24*** join/#asterisk envoy (i=bri@173.sub-75-221-226.myvzw.com)
15:16.30tuxfooTrunk the PBX to * - use what ever interface you want
15:16.57*** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp)
15:17.33tuxfooroute calls between systems via the trunk
15:17.39the_5th_wheelok, that makes sence.So the b410p will act like the teleco Exchange?
15:18.14[TK]D-Fenderthe_5th_wheel : or as a station depending on what kind of port on your PBX you want to plug it into and how you want it to work.
15:18.25envoyWe are using asterisk for our office and for some reason when we try to connect to a conference bridge and enter out conference number the remote PBX doesn't reconize all the digits.  I've adjusted the tx/rx gain in zapata.conf (but all positive values) could that be the issue
15:18.25mintee[TK]D-Fender, thanks for the info... oh, and your well known low-blow too.
15:18.41*** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net)
15:19.16[TK]D-Fendermintee: Wasn't a low blow... that was a "use the right tool for the right job" :)  Someone else would have gone "ouch" if I really threw something at you.
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15:19.29mostyenvoy, analogue telephone line between the two asterisk machines?
15:20.19minteewhatever... it only seems that with you there is a 1-to-1 ratio on answer-insult.  Just saying....   not everyone jumped on the asterisk wagon the day it came out.
15:20.21envoyanalog lines on one asterisk machine (our office) going to customer PBX's
15:20.21[TK]D-Fendermosty: Umm.. think about how that sounds...
15:20.52mosty[TK]D-Fender, i'm asking, not suggesting
15:20.57[TK]D-Fendermintee: s'ok alls good.  just try not to use a spoon as a crowbar and life will be good :)
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15:21.12mintee:P
15:21.14[TK]D-Fenderenvoy: You get some of the digits?
15:21.15mostyenvoy, what dtmfmode settings do you have on your sip clients?
15:21.24envoy[TK]D-Fender, yup
15:21.51mostyenvoy, i think there's a relaxdtmf option in zapata.conf, look at the docs on that and see if it's worth trying
15:21.58[TK]D-Fenderenvoy: Playing with gains is one thing.  try this as well "relaxdtmf=yes" for your channel definition.  You'll have to reload Zap or *.
15:22.28ZaVoidrelaxdtmf? whas that do?
15:22.30envoy[TK]D-Fender, dtmfmode is rfc2833
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15:22.57[TK]D-Fenderenvoy: No need for that yet... you jsut said they are cannected over analog.  DTMFMODE does not factor into this at all.
15:23.20[TK]D-FenderZaVoid: that loosens the constraints on the detection routine when pickup seems spotty
15:23.21*** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83)
15:23.30[TK]D-Fenderenvoy: Also, what ver of * & zaptel are you using?
15:23.44ZaVoidreally...
15:23.49ZaVoidi'm gonna go look at that
15:23.49[TK]D-FenderZaVoid: yup
15:24.28mostytzafrir, where is that patch of yours for a "live" asterisk test install?
15:24.29drakohow is the webpage with the codecs compiled for all plataforms
15:24.35drakoand cpu types
15:24.57envoy[TK]D-Fender, * 1.2.18 and zapata I'm not sure
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15:25.18envoyzaptel rather
15:25.27[TK]D-Fenderenvoy: I would highly recommend upgrading to 1.4.  Zaptel's DTMF detection has undergone MAJOR improvements.
15:26.08funxionAnyone know why I would get dropped calls on a T1 PRI with * v1.2.9.1?  I'm receiving ": Didn't get a frame from channel: Zap/1-1" in debug...
15:26.16envoy[TK]D-Fender, I keep meaning to
15:26.50[TK]D-Fenderfunxion: Check to see if you are getting HDLC aborts, PCI master warnings, of frame slips/losses
15:27.07envoy[TK]D-Fender, any ideas aside from upgrading?
15:27.08funxionI'm not getting any T1 errors oneither side
15:27.16tzafrirmosty, in a bug report:
15:27.27[TK]D-Fenderenvoy: Since it is directly pertinent to your current problem you should perhaps revisit your piority to do so.
15:27.36envoy[TK]D-Fender, or is this a know issue in 1.2?
15:27.46[TK]D-Fenderenvoy: Just gains & "relaxdtmf".  If that doesn't do it, then you know whats next.
15:27.54envoy[TK]D-Fender, thanks
15:28.03[TK]D-Fenderenvoy: Its well known that they did major changes for 1.4 with a reason.
15:28.25[TK]D-Fenderfunxion: is it sharing an interrupts by any chance?
15:28.32funxionnot that I know of
15:28.37funxionbut its possible
15:28.41funxionlemme look
15:28.44[TK]D-Fenderfunxion: Go verify
15:29.57funxiondoesnt seem like it
15:30.00funxionits IRQ 169
15:30.09funxionnever seen a ## that high before
15:30.12funxionqweird
15:31.27[TK]D-Fenderfunxion: That's probably fine
15:32.06ZaVoidoh this is relaxdtmf is only for zapta cards?
15:32.24funxionweird
15:32.31hmmhesays4 hours till my soup is done
15:32.46hmmhesays5 quarts of vegetable roast
15:32.48[TK]D-FenderZaVoid: I'm not 100% sure.... take a look on the WIKI.
15:32.53funxionI just turned on pri debug for the 2 spans and have a test call up waqiting for it to drop
15:32.56ZaVoidi did.. that whats it sounds like
15:33.14ZaVoidoh wait.. Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf.
15:33.16ZaVoidhmm
15:33.16hmmhesays[TK]D-Fender, you ever use a keeley looper?
15:33.29[TK]D-Fenderhmmhesays: Never heard of the term.
15:34.05*** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net)
15:34.16funxionisnt that for music?
15:34.21hmmhesayshardware loop you can put your effect on so you don't have any signal degredation when you aren't using them
15:35.38[TK]D-Fenderhmmhesays: I've seen them on amps (this is really an A/B at that level).  Personally I have no need.  I only run my GT-8, and have no need of any external effects :)
15:36.05[TK]D-Fenderhmmhesays: Guitar > Wireless > Receiver > GT-8 > Amp
15:36.07*** join/#asterisk patrick-- (n=patrick@devnull.biz)
15:36.17patrick--whats the difference between TE and NT mode?
15:36.24hmmhesays[TK]D-Fender, yeah I don't run a gt-8
15:36.29jeremy_gI have installed asterisk 1.4. But now i want to install asterisk 1.6. How do i uninstall asterisk 1.4? I want to remove whatever files were created as a result of ../configure and make install ..everything back to the way it was?
15:36.38hmmhesaysI have a hardware effet loop in my peavey XXL
15:36.42hmmhesaysbut not on my marshall
15:36.51the_5th_wheelWould i be able to connect an ISDN phone to a B410P card?
15:37.09[TK]D-Fenderthe_5th_wheel: Yes
15:37.23the_5th_wheelOk, Cool. Now everything is clear :-D
15:37.42*** join/#asterisk shasta (i=shasta@bluzg.slackware.pl)
15:38.08*** join/#asterisk Spyder12345 (n=bob@169.139.217.48)
15:38.40envoy[TK]D-Fender, Changing my gains fixed the issue.  I had both postive gains and when I lowered them it fixed it.  Any idea on the reason for this
15:39.08[TK]D-Fenderenvoy: Probably to high and was distorting
15:39.11envoy[TK]D-Fender, I would have thought this would be an issue it I was putting negative gains, but...
15:39.45Spyder12345anonyone familiar with tcp dialing via sip in 1.6 or know where I can find some info on this?
15:39.51patrick--anyone?
15:40.15[TK]D-Fenderenvoy: Like on old guitar amps where you up the gain on the pre-amp till it distorts and use the master volume to lower.  Welcome to rock&roll :)
15:40.42[TK]D-FenderSpyder12345: Odds are it'll use whatever protocol you specified for your peer.
15:40.50tzanger[TK]D-Fender: I'm running into two different PBXes now where the PCM data coming in seems awfully 'hot'
15:41.02tzangerif I transmit a ulaw file to the PCM out it sounds fine
15:41.03jeremy_ghow can i undo what  ./configure && make install does to my file system?
15:41.06[TK]D-Fendertzanger: Yup, I've seen a few PRIS like that before.
15:41.14tzangerI don't know if I'm missing a pad on the input or not
15:41.31tzanger[TK]D-Fender: this isn't PRI, this is an actual PBX that I've hijacked the TDM bus from :-)
15:41.49[TK]D-Fendertzanger: O RLY?  Do tell...
15:42.10Spyder12345hmm.. well I am not having much luck with that.. is the peer command to enable it transport=tcp?
15:42.53[TK]D-FenderSpyder12345: You might want to actually follow Mantis, use Google, read docs, etc.... that setting sounds like what I would expect....
15:42.55tzangerI have to play around a little more to see if the the PCM encoder's misconfigured, I've misconfigured a gain stage before it or what
15:43.05ZaVoidmeh i can't find any explanation anywhere what relaxdtmf actually does
15:43.32jeremy_gmake uninstall-all
15:43.34jeremy_gsilly me
15:43.56Spyder12345well none of those places are very useful since I was told this is a new feature only in 1.6. I have already done that with no working results.
15:44.14ZaVoidoo 1.6 does tcp sip?
15:44.37[TK]D-FenderZaVoid: Yup
15:44.41ZaVoidnifty
15:45.09Spyder12345yes it actually listens on both udp and tcp.. But finding good documentation on its use is hard to find.
15:45.52ZaVoidstill can't find anything on relaxdtmf
15:46.11[TK]D-FenderZaVoid: relaxdtmf = picture * "scoring" a tone its receiving and trying to say if I hit 90% accuracy then its OK.  Then relaxed it'll say... you know... 80% isn't too bad now.
15:46.44ZaVoidno i understand that fender.. i'm just look somewhere to see how much it does etc etc
15:47.04ZaVoidbrb
15:48.08*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
15:48.51tzangerZaVoid: it's in dsp.c in asterisk's source
15:50.32*** join/#asterisk fedya (n=fedya@75.112.143.226)
15:52.27BBHosssomeone check this bug out if they have a chance http://bugs.digium.com/view.php?id=11946
15:53.00tzafrirtzanger, hmm... pointing to dsp.c is a very cruel variation of 'use the source, Luke'
15:53.07ZaVoidthanks tzanger
15:56.16ZaVoidhey tzanger i do a more dsp.c | grep relaxdtmf   and i get nothing
15:56.42jeremy_gwhat is that wiki wesbite where people normally search for asterisk info
15:56.45jeremy_gvoip-info?
15:56.49ZaVoidahh case senseitive
15:56.51ZaVoidi found it
15:57.02ZaVoid<PROTECTED>
15:57.22nebojsajsimichow to mark hungup event on any extension ???
15:57.39nebojsajsimiccan it be something like h,1 .....
15:57.40nebojsajsimic???
15:57.42nebojsajsimicor not
15:59.40*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
16:01.19ZaVoidmeh relaxdtmf dind't work very well for digits entered quickly
16:01.34drmessanoso enter them slower
16:01.43ZaVoidyeah i know that
16:01.52ZaVoidbut not everyone will...
16:02.15drmessanoSuccess is only achieved through repeated failure
16:02.24drmessanoThey will learn...... eventually
16:03.13ZaVoidyeah you'd think
16:03.29*** join/#asterisk ManxPower (n=manxpowe@175.sub-75-203-57.myvzw.com)
16:07.04*** join/#asterisk _gm (n=gmustafa@58.27.175.222)
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16:12.27*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
16:12.27*** mode/#asterisk [+o russellb] by ChanServ
16:16.41*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
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16:21.52nebojsajsimicwork if you enable the "*" hangup
16:22.05nebojsajsimichow to enable * hungup
16:23.46*** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net)
16:24.16*** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-33-219.phlapa.east.verizon.net)
16:24.47GrumpManAtWorkanyone using new xml syntax for notifies on polycom phones ?
16:27.17agxwhen i transfer using features.conf is there a channel variable with the number of who started the transfer??
16:28.29*** join/#asterisk fl1p (n=fl1p@port-83-236-208-172.static.qsc.de)
16:28.39*** join/#asterisk lakesolon (n=blake@63.231.182.86)
16:28.55[TK]D-FenderGrumpManAtWork: what "notify"?  Which SIP?
16:28.57ManxPowernebojsajsimic: "core show application dial"
16:29.23ManxPoweragx: if there is it would be documented in channelvariables.txt in the Asterisk source code.
16:29.32*** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100)
16:30.06LakeSolonAnyone know how to disable UPnP in Asterisk? (part of TrixBox)
16:30.10BourrelleOne quick question, if I close a video session with asterisk, does asterisk will send me an ack for the session ending ?
16:30.33hmmhesaysseriously what could possibly cause a guy to shoot his wife in front of kids
16:30.35ManxPowerLakeSolon: UPnP is not part of Asterisk  Ask on the correct channel like *gasp* #trixbox
16:30.39Bourrelleill trying to make the sequence diagram
16:31.02ManxPowerBourrelle: that might be a question for #asterisk-dev
16:31.10Bourrellethx
16:32.21LakeSolonManxPower: well I didn't think it was part of asterisk either, but UPnP is an application-level proto, so it's gotta be asterisk doing it. Either that or it's traversing my NAT by *magic* =p
16:33.05LakeSolonManxPower: I just mentioned it's a trixbox install if that might be a clue as to an unusual default config.
16:33.55ManxPowerLakeSolon: ALL of Trixbox uses unusual configs
16:33.57ManxPower~trixbox
16:33.58jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
16:34.48ManxPowerLakeSolon: Huh?  Asterisk does not support UPnP.  Asterisk has it's own SIP specific NAT features.
16:36.20minteeat the CLI, `database show` shows no sign of the following key, however this code http://pastebin.ca/895147  goes straight to 1000.
16:36.44minteeam i missing something?
16:36.48[TK]D-FenderLakeSolon: That is third-party stuff that has nothing to do with *.  Go use their support resources.
16:37.23agxManxPower, unluckly that files does not that that DIALEDPEERNUMBER and NAME are availble only on blindtransfer while attended trasnfer is a bit broken :-P
16:37.24[TK]D-Fendermintee: pastebin your database dump
16:37.44ZaVoidwhat debug would i need to do capture the DTMF.. using rfc2833.... an rtp debug?
16:38.16[TK]D-Fenderagx: thats because an attended transfer is a new call where the audio is passed off.  a blind transfer alerts the server to take the audio immediately and attach it to the new invite.
16:38.36mintee[TK]D-Fender http://pastebin.ca/895150
16:38.41agx[TK]D-Fender, i see but there is a way to know who started the atxfer?
16:39.05ManxPoweragx: attended transfer is considered a 3-way call.
16:39.08agx[TK]D-Fender, i tried DumpChan() and its quite empty
16:39.17minteeunless it's picking up the /AMPUSER/2957/voicemail, which doesn't match the DID
16:39.18ManxPowerso, whoever the call came from is who did the transfer
16:39.27[TK]D-Fendermintee: I see...
16:39.39[TK]D-Fendermintee: exten => s,1,GotoIf(DB_EXISTS(voicemail/${DNID})?1000:2) <-- because you have fogotten how to EVALUATE a function.
16:39.39ManxPowermintee: Why do you think we can help with a system that was originally setup for AMP?
16:39.54[TK]D-FenderManxPower: nope, obvious dialplan error
16:40.09ManxPower[TK]D-Fender: You're so smrt!
16:40.09funxion[TK]D-Fender can you take a look at http://www.pastebin.ca/895151 its the debug from a dropped call
16:40.28agxManxPower, true but the callerid (strange) is from the original group :-) not the phone that answered the call [ i call 100,Dial(SIP/A&SIP/B), SIP/A atxfer to SIP/C
16:41.16ManxPowerJust to be pedantic, you transfer to an EXTENSION, not a device.
16:41.34*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:42.17ManxPowerfunxion: "Cause: Wrong message (98),"  Looks to me like a signalling issue.
16:42.44funxionbut Im not getting any errors on the circuit
16:42.52ManxPoweryes you are, I just posted it.
16:43.11ManxPowerWhat IS your signaling set to?
16:43.13[TK]D-FenderManxPower: O S I M!
16:43.13mintee[TK]D-Fender, do explain... from GotoIf(condition?label1[:label2]) where label1 = true and label2 = false.... as well as  DB_EXISTS(<family>/<key>) ...  I'm not sure how  GotoIf(DB_EXISTS(voicemail/${DNID})?1000:2)  isn't working.  It was working about 15 minutes ago, until I put the Set() function in there
16:43.26[TK]D-Fendermintee: ${} <-------
16:43.40[TK]D-Fendermintee: You are not evaluating your function.
16:43.44funxionpri_net switchtype=5ess
16:43.57agxManxPower, yes only variable i have is TRANSFERERNAME=SIP/A-083d42b8 ... unless patching res_features.c i see no way of knowing who answered the group call....
16:43.58ManxPowerfunxion: and does it start working if you try setting it to national?
16:44.12ManxPoweragx: You are looking for CALLERID
16:44.24ManxPoweras you are not really doing a transfer, you are doing a THREE_WAY CALL.
16:44.44ManxPowerattended transfers are basically 3-way calls where the person that does the 3-way call drops out of the call
16:44.46agxManxPower, caller id is set to "100" that is the extension that make Dial(SIP/A&SIP/B)
16:44.48funxionManxPower its not that it doesnt work, it dropps calls randomly
16:45.07ManxPowerfunxion: are you going to argue with me or are you going to try my suggestions?
16:45.18funxionsince I've changed the timing source and updated the zaptel version its gotten a little better
16:45.22funxiongoing to change it
16:45.29funxionjsut giving more info
16:45.32ManxPower5ess and national are VERY, VERY similar.  In fact they are interchangable if you want random problems.
16:45.42funxionlol
16:45.59funxionnational would be NI2 in a nortel?
16:46.16mintee[TK]D-Fender, thanks... I got it.  Not sure i understand why, but i'll figure it out.
16:46.27ManxPowerno, national is National ISDN 2, it is a vendor agnostic protocol.
16:46.53agxManxPower, the issue is only when using features.conf inband dtmf; with SIP phones with the transfer button (snom, junkstream, etc.) i don't have this problem the callerid is correct
16:47.28[TK]D-Fendermintee: Because its a function that you have to evaluate. "NoOp(CALLERID(num))" winn NOT show you the caller ID, but rather "CALLERID(num)".  "NoOp(${CALLERID(num)}) will evaulate the functionand return the result
16:47.30cpmpoppr0n!
16:47.35cpmcornpr0n
16:47.38[TK]D-Fendermintee: this is dialplan 101
16:48.16ManxPowermintee: you might want to learn how to use the dalplan 8-)
16:48.50minteeyes, i'm working on asterisk 101..   :D
16:48.54mintee:P
16:49.38[TK]D-Fendermintee: reason it came back as "true" for your gotoif, is because it lokos to see if everything before the "?" comes back as "0".  if it does, then its false.  ANY other value is considered "true"
16:49.42jameswfanyone know of any modern nethdlc docs
16:50.00[TK]D-Fendermintee: Including your unprocessed attemtp to call a function which is jsut a pile of text.
16:50.13minteegotcha, so to eval, it must be within ${}...
16:50.25[TK]D-Fendermintee: Correct.
16:50.32*** join/#asterisk madgeek (n=madgeek@i2router.fi.edu)
16:50.40[TK]D-Fendermintee: the same way you reference a variable
16:50.57minteethanks... yeah, i grasped the idea at first, with the ${NDID} but didn't realize the original function.
16:51.01minteeexactully
16:51.35[TK]D-Fenderabslutly
16:51.36[TK]D-Fender:)
16:51.37ManxPowerNDND?
16:51.57*** part/#asterisk madgeek (n=madgeek@i2router.fi.edu)
16:52.08[TK]D-Fender${DNID}
16:52.16outtolunc${TYPO}
16:52.28mintee${FAIL}
16:52.45outtolunc${KILL}
16:53.00[TK]D-Fender${STFUKPLZTHXBIBI}
16:53.02jameswf~nethdlc
16:58.55*** join/#asterisk barsik76 (n=barsik3@mail.sigmagroup.com)
16:58.58*** part/#asterisk joshkidd (n=josh@adsl-068-209-028-087.sip.asm.bellsouth.net)
17:00.00*** join/#asterisk [T]ank (n=ckwall@206.71.78.158)
17:00.10*** join/#asterisk freezey (n=freezey@gw.mypublisher.com)
17:00.13barsik76hi guys.
17:00.30[T]ankwhat is the preferred monitor type in queues.conf?
17:00.34freezey[TK]D-Fender: got that visio diagram done.. finished it yesterday
17:00.35[T]ankmixmonitor or monitor
17:00.40barsik76i'm trying to setup an asterisk pbx but running into a problem with dnd.
17:00.54barsik76it's not visibile on the fop.
17:01.19barsik76it seems like polycom integrated a new feature in their 2.2 firmware that allows server based dnd.
17:02.02barsik76does anyone know anything about voIpProt.SIP.serverFeatureControl.dnd ?
17:02.20*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
17:03.54*** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
17:08.44[T]ankreading through the docs and other web pages I am not finding the pros and cons to monitor vs mixmonitor. can anyone give me their opinion on it?
17:11.33[TK]D-Fenderfreezey: \o/
17:12.01[TK]D-Fenderbarsik76: * is not able to process that yet
17:12.16funxionhas anyone done a asterisk cluster?
17:12.37[TK]D-Fenderbarsik76: I had a nifty idea on how to implement it, but it hasn't been done to date
17:14.23freezey[TK]D-Fender: so how should we get this over to you? so you could take a look
17:14.34[TK]D-Fenderfreezey: imgshack, etc
17:14.44freezeyk
17:14.46SteveTotarowhy isn't bristuff part of the asterisk source?
17:14.47barsik76[TK]D-Fender Sorry.
17:14.49freezey1 sec
17:15.12[TK]D-FenderSteveTotaro: because its authors didn't or couldn't dislaim it to Digium
17:15.16[TK]D-Fenderdisclaim.
17:15.26barsik76[TK]D-Fender: are there any plans regarding implementation in * 1.6?
17:15.37[TK]D-Fenderbarsik76: I already answered that.
17:16.04*** join/#asterisk AndyGraybeal (n=andy@node191.34.251.72.1dial.com)
17:16.19SteveTotaroi find it hard to believe they would not disclaim patches
17:16.40[TK]D-FenderSteveTotaro: Maybe they use other GPL code they have no rights to <-
17:16.52SteveTotaroyeah, i suppose
17:16.58*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
17:17.01SteveTotarocrappy GPL
17:17.34SteveTotarofree as in beer I say!
17:18.04Qwelldoesn't bristuffed cause a lot of issues as well?..
17:18.18barsik76[TK]D-Fender: I'm new to asterisk I'm not sure what is Mantis
17:18.47[TK]D-Fenderbarsik76: the bug tracker that tracks new pattches, etc
17:19.08Qwellbarsik76: bugs.digium.com
17:20.52*** join/#asterisk shinao1 (n=shinao1@80.248.0.59)
17:21.20SteveTotaroi don't know but i just spent the last 24 hours fighting with bristuffed
17:21.38SteveTotaroproblem was the upgraded kernel
17:21.57SteveTotarosystem clock was jumping all over the place
17:22.28SteveTotarotzafrir is my hero ;)
17:22.33*** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net)
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17:26.41bsdwarriortkd-fender Im really stuck with setting the userfield  for cdr in a perl script. can you point me in the right direction
17:28.16[TK]D-Fenderbsdwarrior: set it before you do your outbound dial for your first leg of the call.
17:32.15*** join/#asterisk RoyK (n=roy@ip-107-15-149-91.dialup.ice.no)
17:32.19patrick--Hey i keep getting this error:
17:32.21patrick--Got EVENT_FACILITY but we don't have a ch!
17:32.30patrick--how can i fix it?
17:32.41patrick--its a chan_misdn error
17:32.42bsdwarriortkd-fender so send a separate action command and set the variable ?
17:33.16[TK]D-Fenderbsdwarrior: You have to set the FUNCTION, which means you have to do it in the dialplan.
17:33.38outtoluncjust add 'Variable: __VAR=VALUE' to the bottom of the rest of the originate
17:33.39[TK]D-Fenderbsdwarrior: CDR is not a "variable".  That tells you you have to be executing dialplan code to set it.  Think on that
17:34.04outtoluncthen you can set the cdr(yadda) to it on the way back in
17:35.30*** join/#asterisk drmessano-LT (n=nonya@207.230.140.240)
17:38.26patrick--Hey i keep getting this error:
17:38.28patrick--Got EVENT_FACILITY but we don't have a ch!
17:39.00[TK]D-Fenderpatrick--: And we only heard you ask that 5 minutes ago...
17:39.07patrick--sorry
17:41.04patrick--google wont give me no answer
17:41.13outtoluncThis may come from a call we don't know nothing about, so we ignore it.
17:41.47patrick--mhh, well its to an extension mentioned on the dialplan
17:42.29outtolunchaving an exten in a dialplan does not a channel make... until you start a pbx event on it
17:43.01patrick--outtolunc: how do i "make" a channel? i though my misdn was setup correctly
17:43.20b11d|bblDid the behaviour of Comedian Mail change at all between 1.2 and 1.4?
17:43.55outtolunceither you are are attempting to do something (like a hint/etc) on a channel that is not created yet, or has already gone bye bye
17:44.06outtoluncwould be my guess
17:45.30b11dI am finally moving my PBX to 1.4 and would really like to know if my end users will notice any difference in their voicemail.  I dont currently see any changes, but would like to know for certain.
17:45.43b11dchanges.txt doenst list anything that I can see..
17:45.51barsik76[TK]D-Fender: i don't seem to be able to find any information regarding serverFeatureControl.dnd on bugs.digium.com
17:45.52outtoluncactually this is a facility event.. so it would be a network issue
17:45.54b11derr upgrade.txt
17:46.06*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:46.08barsik76[TK]D-Fender: is it under some different name?
17:48.48[TK]D-Fenderbarsik76: I told you it doesn't EXIST.  It is not supported yet.  You can stop looking.
17:49.08RoyKhow does asterisk compare to this? http://www.museumoflondon.org.uk/piclib/images/%5CMID%5C0330001612_5mb.jpg :D
17:49.41drmessano-LTAsterisk takes the people out of the loop... the weak link :)
17:50.25RoyKhehe
17:50.37b11danyone? any changes to the voicemail?
17:50.43b11dI will assume none..
17:54.20*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
17:55.47*** join/#asterisk rcahilig (n=test@203.115.187.98)
17:56.19[TK]D-Fenderb11d: nothing really.  The recordings are a different picth because of their rebuild
17:56.36b11das long as the behaviour didnt change, thats all I care about..  Thanks TK
17:57.07[TK]D-Fenderb11d: No, functions the same.
17:57.09rcahiligHi, How do I install PHPAGI, I cannot find any tutorial on google installing PHPAGI
17:57.24[TK]D-Fenderrcahilig: ...
17:57.26[TK]D-Fender~book
17:57.27jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
17:58.36[TK]D-Fenderrcahilig: http://phpagi.sourceforge.net/
17:59.02*** join/#asterisk klictel (n=klictel@atelka.info)
18:00.23b11dwhile im thinking of it, is there any way to enforce a voicemail password policy? I'd like users to have to change their VM passwords every now and again..
18:00.59[TK]D-Fenderb11d: You could do this with excessive dialplan scripting, but nothing convenient.
18:02.48b11dah.. its probably overkill anyways.
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18:22.58jeremy_gwhats this -lwrap library that asterisk 1.6 needs?
18:24.18*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
18:24.28na0miHi I just installed asterisk last night and am very happy with it - however something strange has happened - I was able to recieve externel calls through my voip provider but now I get this message and the call clears - but I swear I never changed a thing - any ideas? http://pastebin.ca/895294
18:27.47keith4na0mi: codec negotiation problem?
18:28.05jeremy_gdo i need to repeat myself to get answered?
18:28.13keith4what codecs are you accepting?
18:28.15patrick--How do i create a channel for my HFC cards to be able to communicate at?
18:28.22keith4jeremy_g: no, but you might have to be patient
18:28.23na0mikeith4: have allow all in and it did work
18:28.32*** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar)
18:28.50*** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net)
18:29.21barsik76[TK]D-Fender: is it possible to use presence to view whether the user's phone on dnd?
18:29.22keith4jeremy_g: presumably it's lib wrap ...
18:29.29keith4TCP wrapper library
18:29.46keith4are you asking a trick question?
18:30.32*** join/#asterisk clandmeter (n=Carlo@81.175.82.2)
18:31.50[TK]D-Fenderbarsik76: No
18:31.51na0milol - just got an email from my voip provider they have problems their end - guess its that
18:33.51patrick--can anyone tell me how to setup channels with misdn?
18:34.39barsik76[TK]D-Fender: is there a way to view user's dnd status with polycom phones without using the *78/*79?
18:35.20barsik76[TK]D-Fender: i would like the phones to dispay if it's on dnd or not...
18:38.42*** join/#asterisk moellerdk (n=chatzill@4806ds1-hl.0.fullrate.dk)
18:40.12jameswfjbot: tell patrick-- about wikis
18:40.17hmmhesaysdoesn't dnd change the subscription status?
18:41.04plikna0mi: Hello :)
18:41.13na0miplik: Hi
18:42.00[TK]D-Fenderbarsik76: No.
18:42.13[TK]D-Fenderhmmhesays: No.
18:42.44hmmhesayshmm it should
18:44.36[TK]D-Fenderhmmhesays: No, DND is a transparent thing that only causes a reject and with a reason code based on provisioning.
18:44.37kyron~book
18:44.38jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
18:44.43kyronSynoptic, ^^^
18:44.48kyronSynoptic, read chapter 5
18:45.05hmmhesaysgotcha
18:52.50*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net)
18:58.26*** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com)
19:00.33*** join/#asterisk murdmath (n=vircuser@17.sub-70-193-177.myvzw.com)
19:07.27*** join/#asterisk servergod (n=servergo@70.97.159.120)
19:07.42servergodhello all!!!
19:09.19BBHosssup dog!
19:10.11servergodtrying to set up asterisk as a B2BUA.  have 1 trunk going voip to cisco gateway to pstn, and another is another asterisk.
19:11.06*** join/#asterisk fl1p (n=fl1p@ip-90-186-3-96.web.vodafone.de)
19:11.18servergodi need for asterisk to send call to B2BUA, then have that one run it through the outbound routes, then out the other sip trunk
19:11.35servergodbut the B2B
19:11.37[TK]D-Fenderservergod: what is this other B2BUA?
19:11.41servergodasterisk
19:11.54servergodsends the calls to s|1
19:12.05[TK]D-Fenderservergod: You are thowing terms an conenction info around without making much sense.  Draw a picture.
19:12.18servergodk brb
19:15.02*** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net)
19:15.14drmessano-LTCan I integrate * and OU812?
19:16.54[TK]D-Fenderdrmessano-LT: Sure.... eat'em & smile :p
19:17.22[TK]D-Fenderdrmessano-LT: Let me know when you've found the reference :)
19:17.41drmessano-LThmmm
19:18.10drmessano-LTThe eat'em and smile isn't obvious to me lol
19:18.26[TK]D-Fenderdrmessano-LT: its both together, and I've dropped other clues...
19:18.56drmessano-LTWell, if it's more than just a vague Van Halen reference, i'm lost
19:20.44[TK]D-Fenderdrmessano-LT: When David Lee Roth left VH in 1985 to go solo his first release was named in revenge "Eat'em And Smile".  When Sammy Hagar filled his place with VH, they responded in kind with the title of their next rease "OU812"
19:20.52[TK]D-Fenderdrmessano-LT: Mutual snub :)
19:21.24drmessano-LTAhhh... Little trivia I had no clue about.. cool
19:22.11[TK]D-Fenderdrmessano-LT: Another useful tidbit for you....
19:22.30[TK]D-Fenderdrmessano-LT: I used to play with a VH cover band, and did DLR's solo stuff too...
19:22.34*** join/#asterisk southtel (n=southtel@68-114-17-226.dhcp.gwnt.ga.charter.com)
19:22.42drmessano-LTOh cool
19:23.35drmessano-LTI've actually had to forcibly boycott Van Halen
19:24.15drmessano-LTWould you believe that on TWO occasions, I have had two seperate cars break down on me while listening to "Running with the Devil"
19:24.49[TK]D-Fenderdrmessano-LT: You should boycott cars.  Twice they've broken down on you while being on the road!
19:25.00[TK]D-Fenderdrmessano-LT: Go bike-boy!
19:25.53drmessano-LTLOL
19:25.59drmessano-LTWell, they were shit cars
19:26.20drmessano-LTI learned that a car payment is a reasonable tax to keep on the road
19:26.34drmessano-LTBut I couldn
19:26.42drmessano-LTBut I couldn't ignore that coincidence
19:26.51drmessano-LTEddie was speaking to me
19:26.54drmessano-LTChanneling me
19:27.12drmessano-LTKILL KILL KILL, HE SLEWED WITH HIS AXE
19:27.12[TK]D-FenderI just lost a car this week (1998 Taurus SW shitbox), and replaced with a very nice 1998 Chevy Malibu (soptless)
19:27.27*** join/#asterisk emmix-devin (n=devinsai@c-68-51-54-72.hsd1.ar.comcast.net)
19:27.34[TK]D-Fenderdrmessano-LT: if you want "Eddie" to channel.... thats what Iron Maiden is for :P
19:27.42drmessano-LTheh
19:27.47J4k3doh-mestic
19:27.48emmix-devinany here used mediatrix 1124 boxes
19:27.49[TK]D-FenderSpotless even.
19:27.57[TK]D-Fenderemmix-devin: I have, as have others.
19:28.18emmix-devinhow was the quality of the box
19:28.54[TK]D-Fenderemmix-devin: Decent, friendly to use, typical features.  How many ports do you need?
19:28.58drmessano-LTSo [TK]D-Fender, I MUST ask..
19:29.21emmix-devin160 ports
19:29.31drmessano-LTAre you OK with Van Hagar, or did Van Halen die when DLR left?
19:29.31[TK]D-Fenderemmix-devin: Ok, not a bad idea.
19:29.52[TK]D-Fenderdrmessano-LT: Mixed.  Wasn't the same, but Gary Cherone KILLED VH.
19:30.09[TK]D-Fenderdrmessano-LT: DLR si supposed to be working on a comeback album with them right now.
19:30.27[TK]D-FenderGary Cherone = good with "Extreme".... but not VH
19:30.38drmessano-LTI agree.. I liked Hagar.. it wasn't AT ALL the same, but still good.. Gary Cherone just....
19:30.45[TK]D-FenderWRONG!!!
19:30.51drmessano-LTToo much Cocaine...
19:30.51[TK]D-Fenderlol
19:31.04emmix-devinso you think mediatrix is the way to go for that many analog ports, we have used Xorcom and Channel banks, but have not been happy with them
19:31.05drmessano-LTRotted their brains on that decision
19:31.09tnt_#`Has anyone the guide for the 3.0.0 fw for Polycom phones ?
19:31.18[TK]D-Fendertnt_: www.polycom.com
19:31.42[TK]D-Fenderemmix-devin: Yeah, for that kind of density, sure
19:31.54[TK]D-Fenderemmix-devin: Zaptel FXS = ASS
19:31.57drmessano-LTForget THEIR decision, what made Gary Cherone think he could rock out, Van Halen style?
19:32.13[TK]D-Fenderemmix-devin: This way you can have redundency and reduced overhead.
19:32.19drmessano-LTA; More drugs
19:32.27tnt_[TK]D-Fender: nope ... The latest version of their site is 2.2 ... I saw the 3.0 existed only a brief moment when they put the fw by mistake ...
19:32.29J4k3I'm not a big sammy hagar fan, but I'd kill david lee roth given we were in the same room
19:32.34emmix-devinthanks
19:32.35[TK]D-FenderE; Even more drugs still...
19:32.51[TK]D-Fendertnt_: the admin guide is on their site.
19:33.09scooby2any idea how to make zaptel not cause a kernel panic on smp kernels with one cpu?
19:34.51tnt_scooby2: not using a smp kernel ?
19:35.19scooby2almost all linux distributions come with smp kernels these days
19:36.04ManxPowerscooby2: you have some OTHER issue.  Zpatel does not kernel panic just because you have an SMP kernel on a UP system
19:36.33*** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif)
19:36.34tzafrirscooby2, hmmm... can you reproduce such a kernel panic? what kernel?
19:36.46tzafrirwhat hardware involved? what version of Zaptel?
19:36.50scooby22.6.18-53.1.6.el5
19:37.08scooby2zaptel 1.4.8
19:37.11scooby2Dell 1850
19:37.23servergodgot it
19:37.27ArchSSMany analog/bri/pri zaptel cards?
19:37.27scooby22.8ghz xeon w/ 1gb ram
19:37.39scooby2te212p
19:37.55servergodk here is the pic i drawded http://foshizzlenet.net/myspaceimg/b2bua.png
19:37.57ArchSSMand it reboots when you load the module? ... or?
19:38.11scooby2after about 20-25 calls it kernel panics
19:38.41*** part/#asterisk [T]ank (n=ckwall@206.71.78.158)
19:38.43scooby2trying to move from sangoma to this new card
19:39.15J4k3its an old dell, the machine is probably just getting tired.
19:39.25[TK]D-Fenderservergod: Why 2 * boxes?
19:40.47scooby2i can confirm it on 2 identical machines
19:41.01servergodwant to use the b2bua as a main gateway for all of our devices. We have about 6 customers with 2651xm that feed pri to vendor pbx's.
19:41.31servergodeach of those need full dial plans. it would just be easier to send all calls to the b2bua,
19:41.52[TK]D-Fenderservergod: Ok, so the 2811 does what exactly?
19:42.47Qwellservergod: drawded?
19:43.02BadHorsiecan i call ChanSpy several times over the same channel at the same time?
19:44.04tzafrirscooby2, any chance you have a trace from that panic?
19:44.42*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca)
19:45.07*** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com)
19:45.40[TK]D-FenderBadHorsie: I think there may be an opening for you at AT&T
19:45.55drmessano-LTROFL
19:45.59drmessano-LTpwn3d
19:46.16*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:46.17*** join/#asterisk maszlo (n=reckenro@65.223.240.146)
19:46.33jameswfI haer AT&T voted dfor ron paul
19:46.52jameswfstupid korean keyboard
19:46.55maszloi am having problems getting the callerID to get sent along when having a line setup on call foward
19:47.03maszloThe fowarded call carrys the callerid of an internal number, what can i do to make it the actual caller?
19:47.36drmessano-LTAT&T voted for Ron Paul
19:47.41drmessano-LTSo they're the "one"
19:47.44jameswfmaszlo: what is your trunk type
19:47.59maszlosip
19:48.20maszloits coming from a pri
19:48.22jameswfsome sip providers and ron paul do not allow caller ID spoofing
19:48.42maszlolol
19:49.02J4k3~ron paul
19:49.03jbotZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT
19:49.04BadHorsie[TK]D-Fender, sorry for trying to learn asterisk :P
19:49.20jameswfBadHorsie: is forgiven
19:49.28jameswfnow run away
19:49.31jameswf:)
19:49.34hmmhesaysatlantis flight looks good
19:49.58[TK]D-FenderBadHorsie: Just that you're trying to seriously gang-rape your callers.  Like... WTF?
19:50.34maszlowhere should i start to figure out if it is the provider to blame?  its verizon, we were able to make calls on the pri before we even had the numbers ported, w/o a callerid
19:50.37J4k3good sip providers don't care, since they know they're used for termination for numbers that they don't own.
19:51.04maszloi think that it is something in the configureation, i am just new with this system
19:51.12maszloglad it makes calls ;)
19:51.38jameswfFor a while I was calling my brother with his cell phone... I tried calling his sellphone and got in to his voice mail t-mobile rocks
19:52.20maszloi think most cell phones are like that
19:52.50maszloor were you spoofing the caller id to get into his voicemail?
19:52.53[TK]D-Fenderone piece of happy news today.... looking like Mitt's dropping out of the race...
19:53.07drmessano-LT[TK]D-Fender: How can repeatedly ChanSpy on a caller, record their convo, and then e-mail it to our CEO and CC our personal GMAIL inboxes?
19:53.20drmessano-LTYAY
19:53.27[TK]D-Fenderdrmessano-LT: Forgot the cherry on top!
19:53.46drmessano-LToh
19:54.03drmessano-LT[TK]D-Fender: How can we repeatedly ChanSpy on a caller, record their convo, and then e-mail it to our CEO and CC our personal GMAIL inboxes AND post it on YouTube?
19:54.24drmessano-LTShit, left out MySpace
19:54.27[TK]D-Fenderdrmessano-LT: And no, not e-mailing... those dinosaurs force other concessions like warranting automatic "burning" of the recodings on LP and snail-mailing them.
19:54.36drmessano-LTROFL
19:54.48[TK]D-Fendernow THAT pwns
19:55.02drmessano-LTyes, yes it does
19:55.32drmessano-LT"IIIIII can't lie.... it's FOURTY FIIIIIVE"
19:56.07J4k3[TK]D-Fender still listens to wax tube recordings.
19:56.28drmessano-LTI want one of those Laser LP players..
19:56.35J4k3me too
19:56.53J4k3I recently had opportunity to get a couple albums (same album, two copies) ripped with one
19:57.04maszlocan i get a direction to where i should look to straighten the callerid, actaully they are all getting the same number, all 15 phones make calls with the same outbound callerid, so i guess that means that spoofing is allowed
19:57.11J4k3trying to make a high quality copy of an album whos sources got destroyed 25+ years ago
19:57.20drmessano-LTAh
19:57.54*** join/#asterisk CVirus (n=GoD@82.201.174.232)
19:57.57drmessano-LTWe still have turntable in one of our production rooms here.. I haven't put a stylus on it in years lol
19:58.18drmessano-LTI'm willing to bet the motor doesn't work
19:58.22J4k3haha.  a stereo isn't a stereo without a turntable. ;)
19:58.39J4k3I *hate* CDs... they're a fucking high dollar ripoff of lousy quality and horrible longevity
19:59.05drmessano-LTPurple Haze on LP.. it just get's NO better
19:59.12drmessano-LTgets*
19:59.54drmessano-LTI listen to the Beatles on CD and there's no comparison to the LP's I grew up with
20:01.13J4k3well, lots of music just doesn't sound right otherwise...  for example heart - dreamboat anne - magic man... it sounds completely different off a digital source because you're *supposed* to hear the tonearm resonance
20:02.09J4k3it may not have been engineered to sound that way, but thats the way everyone got to hear it for like 15 years... then CDs come along and change it.
20:02.09maszlorecord players are nice, but headphones are a key part of the equation as well
20:02.18drmessano-LTYes.. and listening to any of the Doors CDs is the same way...
20:02.18J4k3but I'll be the first to say 33.3 was too damned slow
20:02.29J4k3what this world needed was more 45 rpm 2-disk releases
20:02.50*** join/#asterisk javar (n=javar@69.79.134.24)
20:02.55maszloi got a pair a akg muffs last month best set i have ever owned
20:02.56J4k3yeah... gotta have a good set of headphones for any musical enjoyment
20:03.02*** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-03c2ea82100cd452)
20:03.38J4k3I got a pair of cheapish sennheiser closed-back
20:03.40b1ch0drmessano, are you there ??? need help again !!!!
20:03.41J4k3pro 280s I think
20:03.47drmessano-LTI am here
20:03.58b1ch0how can i change default language (en) to es ?
20:04.07J4k3they're not perfect, but they effectively sound better than my previous pair (HD600's) due to not letting in all the background noise.
20:04.28maszlothey noise canceling or something?
20:04.35J4k3nah, just fully closed-back
20:04.49J4k3about 30 db of natural rejection
20:05.00drmessano-LTlanguage=es
20:05.33*** join/#asterisk [hC] (n=hardcore@vpn.voxter.com)
20:05.49maszloyeah its all about that akg k240
20:05.56b1ch0ok, but where do i change it ? iax.conf, sip.conf , zapata.conf
20:06.56drmessano-LTsip.conf at the very least
20:07.01drmessano-LTNot sure about the others
20:07.10*** part/#asterisk beek (n=klinebl@65.211.106.243)
20:08.39*** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo)
20:10.29*** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net)
20:10.49maszlocallerid?? can someone tell me where to adjust this?
20:10.55*** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta2 (2008/01/28), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox
20:11.09drmessano-LTYAY
20:11.20*** join/#asterisk Strom_M (n=strom@208.127.172.112)
20:12.06drmessano-LTWebsite hasn't been updated
20:12.14drmessano-LTor I am in cache hell
20:14.02*** join/#asterisk servergod (n=servergo@70.97.159.120)
20:15.14jameswf1.4.18 is release?? why dont i see that on the lists
20:15.26Qwellbecause you haven't looked
20:15.48Qwell(that, and they probably haven't been sent out yet.  give the man a minute)
20:15.57jameswfno NOW :)
20:16.08Qwellk/ickban jameswf nub
20:16.09Qwellerm
20:16.33Qwell:D
20:16.47drmessano-LTIt's about time.. What happened to monday night???   I WANT MY MONEY BACK
20:17.49jameswfoooh they are offering refunds now....
20:17.57J4k3"the boys in brown" takes on a whole new meaning.
20:18.11*** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170)
20:18.36LeBowlingAlleyhas anyone seen a problem of a Parked Call answering an incoming call?
20:18.47QwellLeBowlingAlley: you're gonna need to be a little more specific
20:18.52drmessano-LTNow that 1.4.18 is out, HappyClownPBX is getting a refresh
20:19.14drmessano-LThonk!
20:19.23jameswfI realy need to start work on xobxirt
20:19.36drmessano-LTyes you do
20:19.50drmessano-LTBefore I make HappyClownPBX public
20:20.08LeBowlingAlleyA call over a zap channel came in and rang, and then eventually connected to a call that was in the parking lot
20:25.55LeBowlingAlleyQwell: http://pastebin.ca/895421
20:26.17LeBowlingAlleyThe parked call answering the incoming call over a zap channel is at the end
20:26.23*** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr)
20:28.09[TK]D-FenderLeBowlingAlley: Yay, one of your callee's forwarded their phone to "71" picking up a parked call.  Go fix their PHONE.
20:28.31Qwellor fix the user
20:28.39LeBowlingAlleyIs that the 209 user?
20:29.46maszlowhere do you set an outbound callerid?
20:29.52*** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr)
20:29.57jhiverhi guys
20:30.14jhiverdo you know a way to batch convert a bunch of .wav files to .g729 format?
20:30.35Qwelljhiver: use asterisk's convert CLI command
20:30.35[TK]D-FenderLeBowlingAlley: Feb  7 15:10:36 VERBOSE[20561] logger.c:     -- Got SIP response 302 "Moved Temporarily" back from 192.168.4.109
20:30.45[TK]D-FenderLeBowlingAlley: Feb  7 15:10:36 VERBOSE[9866] logger.c:     -- Now forwarding Zap/5-1 to 'Local/71@from-internal' (thanks to SIP/209-09146bf8)
20:30.49[TK]D-FenderLeBowlingAlley: What do YOU think?
20:30.57Qwelland a simple for loop with asterisk -rx - assuming you have g729 codec licenses
20:30.58LeBowlingAlley:D  Thanks.
20:31.00*** join/#asterisk beighto (n=chatzill@12.176.156.130)
20:31.25[TK]D-Fenderjhiver: Digium's site has a converter as well.
20:31.36Qwell[TK]D-Fender: really?
20:31.36[TK]D-Fenderjhiver: depending how many you need to do.
20:31.42*** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp)
20:31.43[TK]D-FenderQwell: Used to...
20:31.45*** join/#asterisk RoyKa (n=roy@ip-77-15-149-91.dialup.ice.no)
20:31.50jhiverabout 150
20:31.54QwellI know we did at one point, but I thought that was for prompts from thevoice
20:32.32*** join/#asterisk servergod (n=servergo@70.97.159.120)
20:33.01Qwelljhiver: do you have g729 licenses for asterisk?
20:33.32jhiverso first i batch convert to 8khz using sox, then i write a shell script to use asterisk -rx 'file convert foo.wav foo.g729'
20:33.43jhiverwell i can always buy ONE
20:33.51jhiverthat won't kill me
20:33.54J4k3you need g729 licenses, then you gotta find g729 codecs that actually work :P
20:34.57hmmhesaysmccain is staying pretty cool at his speech
20:35.46jhiverbeside it looks like i have the g729 hx0rd codec on the box, oops
20:35.50jhiverwhich is crazy since i don't transcode at all on this box :)
20:36.09[TK]D-Fenderhmmhesays: Yah, while C&L just said that if he makes it to office he'll make Cheney look like GHANDI
20:36.36[TK]D-Fenderhmmhesays: Mccain = militant.
20:36.45J4k3mccain = old gay turd.
20:37.10jhiverso will asterisk's 1.4 file convert utility be smart enough to convert to 8khz or do i need to do it first?
20:38.15hmmhesays[TK]D-Fender, militant not so much
20:38.24hmmhesayswe're fscked if hillarybot3000 gets in
20:39.00jhiveraah politics :(
20:39.01[TK]D-Fenderhmmhesays: This is Mr. "in Iraq for the next 100 years, and other wars to come"
20:39.09*** join/#asterisk lgc (n=lgc@189.146.43.17)
20:39.36hmmhesaystaking that statement way out of context
20:39.39jhiverhere in france we have been fscked for years, and sarkozy is just the latest fuckage in a long line of fucked up presidents :)
20:39.49lgcHi. What do I need to use Asterisk in order to send faxes from my computer?
20:40.01hmmhesaysuse callweaver
20:40.03[TK]D-Fenderhmmhesays: Yes, Hillary is bad, so is every republican except Ron Paul who if he doesn't pick up his act FAST had better run as a Libertarian Party candidate for the final
20:40.22hmmhesaysRon Paul I would vote for if I knew he was going to get in
20:40.36hmmhesaysyou know obama talks a big game but he'll never follow through
20:40.44J4k3Ron Paul is the sucker candidate
20:41.02lgchmmhesays, was the callweaver answer for me?
20:41.02J4k3he's the person the republicans are using to get people with an IQ over 80 to vote republican.
20:41.07hmmhesayslgc, yes
20:41.10[TK]D-Fenderhmmhesays: He will probably not get anywhere as a Republican, but he can make a place on the Ballot that counts, but he's so conservative noone knows who he is!
20:41.19J4k3he has no chance, nor does his party ever plan to give him a chance, to do anything useful
20:41.31lgchmmhesays, let me check callweaver...
20:41.51hmmhesays[TK]D-Fender, he is. I think that that mccain will be the best candidate, sad but true. He is a fiscal conservative which is what we need right now
20:42.00[TK]D-FenderJ4k3: Latter sure, former depends... there is a small chance.  But if Barack makes the Dem side I could live with that.
20:42.19hmmhesayswe're just as fscked if obama gets in
20:42.19J4k3[TK]D-Fender: hopefully he will..  I don't think I can vote for hillary clinton.
20:42.30hmmhesaysWe don't need more government we dont' need more social programs
20:42.41J4k3we don't need iraq
20:42.43hmmhesayswe need less social programs,  free market health care
20:42.50[TK]D-Fenderhmmhesays: How can you say fiscal when the dumbass is ready for war with the world and to maintain forces in Iraq for the nexyt CENTURY?  THERE'S your econimic sink-hole
20:42.51J4k3social programs don't cost the US money, it rolls right back in
20:42.54jameswfif you hate babies and wanna kill all small animals vote hillary
20:43.00J4k3assuming your social programs don't involve buying a lot of foreign items
20:43.07[TK]D-FenderJ4k3: No, she is as bad as the worst of them.
20:43.20hmmhesaysthe democrats have a great vision, but they are fiscal morons
20:43.32hmmhesaysin general
20:43.39[TK]D-Fenderhmmhesays: Except for Kucinich who dropped out...
20:43.42J4k3hmmhesays: yes and no...  step 1 to fisical stability - don't start an empire you can't maintain.
20:43.47jameswfI heard hilliry likes to kick small kids with down syndrome
20:43.57hmmhesaysI heard hillary is the man in bed with bill
20:43.59[TK]D-Fenderhmmhesays: Barack isn't too bad.  by pulling out of Iraq he'd have a lot mroe to work with
20:44.07hmmhesayswe can't just pull out of iraq
20:44.15hmmhesaysit needs to be calculated with set goals
20:44.15J4k3sure we can
20:44.17jameswfreal men dont pull out
20:44.23J4k3we ran up in there without any plans or justification
20:44.26J4k3we can leave just the same
20:44.41J4k3real men get done and go to bed
20:44.41*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
20:44.59jameswfwe still hvent puled out of any place we have fought a war except vietnam and look what happened there
20:45.18[TK]D-Fenderjameswf: Never really pulled out.... still have permanent bases...
20:45.19drmessano-LTIf McCain wins, we're going back to Vietnam to finish them off!
20:45.22J4k3jameswf: vietnam?  happened?  we just fucked over those that sided with us... and that should be a good lesson to the rest of the world.
20:45.29hmmhesaysdrmessano-LT, bs
20:45.39husimonin zt monitor what should the levels be, about 50%?
20:45.48jameswflol you think bush jr had a vendetta
20:45.54J4k3we're as bad as the french, except we have more corporate evilness built in.
20:45.57*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:45.57drmessano-LTYep.. and we didnt go into Iraq for Oil and "He tried to kill my dad"
20:46.08drmessano-LTwhatever
20:46.23J4k3"he tried to kill my dad"... and we thanked him for it and was saddened that he didn't get the job done.
20:46.25drmessano-LTMcCain is gonna send us back to Vietnam, like a friggin Chuck Norris movie
20:46.32J4k3too bad nobody kicked barbara in the stomach a few dozen times :P
20:46.52drmessano-LTMISSING IN ACTION 6: HERE COMES THE MARINES
20:46.56jameswfyou know the iraq war is about oil but if you own a car or anything delivered anywhere by a vehicle stfu cause it doesnt go without oil
20:46.57husimonjameswf,  I think hating babies and small animals is the least of our worries right now :P
20:47.19J4k3jameswf: it ran on much cheaper oil before the war.
20:47.25drmessano-LTTrue
20:47.26J4k3jameswf: I didn't need that war to keep BUYING oil
20:47.52J4k3and that oil was being pumped much cheaper by saddam via france, than it is the US contractors.
20:47.58husimonso the war wsas to what, protect oil prices?
20:48.03drmessano-LTWe went over there to librate them of their existance
20:48.03J4k3hence why oil went from $23 to $105 a barrel.
20:48.06husimonor to protect us oil p
20:48.11J4k3husimon: keep oil prices high
20:48.19J4k3husimon: same as the war on drugs... the war on drugs is to keep the profits high.
20:48.19jameswfwell if we got rid of all this deplomicy crap and leveled iraq and made it a giant golf course for the US oil would be like a buck a barrel
20:48.27husimonJ4k3, ah ok i was going to say no fucking way the war was to make it lower
20:48.31J4k3husimon: if you could buy cocaine in stores, it'd cost about as much as talcum powder.
20:49.01husimonJ4k3, i think the manufacturing process might be little more then talcum powder :P
20:49.09J4k3and you wouldn't have crackheads kicking in your grandmother's door to rob her for the last few cents off her pension check.
20:49.20husimonbut I get your point
20:49.23J4k3husimon: not really, and its that cheap columbian labor
20:49.24J4k3;)
20:49.32husimonyou still need gas
20:49.36jameswfI would like to see all the trupes pulled out of iraq and dropped in ti vietnam but the truth is no matter the president we will probably stay.... even billary wont commit ti a timeline
20:49.42husimonfor cocaine, so maybe this war was a war on drugs
20:49.45husimontwo for one
20:49.49J4k3haha
20:49.56J4k3gas for cocaine?  just ether.
20:50.06J4k3well, and you gotta cook it down
20:50.08jameswfdamn i cant type... damn korean kyboard
20:50.09*** join/#asterisk pkunkra (n=chris@cpe-72-229-148-29.nyc.res.rr.com)
20:50.26drmessano-LTJohn McCain "Elect me, and I will win the Vietnam War... for the POWs, for Chuck Norris, and for Rambo!"
20:50.42Qwelldrmessano: Chuck is with Huckabee
20:50.47Qwell(seriously)
20:50.53drmessano-LTOh thats right
20:50.58drmessano-LTHe's devout and all
20:51.05jameswfgod i saw the new rambo and holy crap there is a movie that should have never happned
20:51.07drmessano-LTWWJVF
20:51.07J4k3yeah, huckabee is the only guy thats a big enough asshole to keep him around.
20:51.10Qwellthey were here in Huntsville last week :p
20:51.11drmessano-LTWho Would Jesus Vote For?
20:51.19J4k3man, huckabee is a piece of shit to speak to in real life.  total prick.
20:51.41drmessano-LTSo McCain is gonna go back for the POW's and John Rambo
20:51.47drmessano-LTNever forget!
20:51.50J4k3maybe we can turn mccain back into a POW
20:51.56J4k3could we, like, give him back?
20:52.04jameswfone thing you can say about billary is she doesnt have a prick.... or can you
20:52.11J4k3well
20:52.20J4k3the only thing I can say about hillary is you don't have any fear of her sucking any corporate dick
20:52.24J4k3she won't even suck her husband's
20:52.31Qwell#politics
20:52.33Qwellkthx
20:52.36drmessano-LTWell, with Billary, the white house will always be fully staffed by women
20:52.50J4k3topless women ftw
20:52.56drmessano-LTboobies
20:52.58J4k3well, as long as hillary keeps hers on
20:53.04jameswf~boobies
20:53.04jbotwell, boobies is (.)(.)
20:53.05J4k3that'll turns my smile upside-down.
20:53.19*** part/#asterisk rcahilig (n=test@203.115.187.98)
20:53.54husimonhow can I tell what zaptel channel my call is being made on?
20:54.22drmessano-LTNot that I enjoy installing XP all that much.. but I can't wait for SP3.. Damn 3 hours of updating from SP2 level
20:54.22*** part/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
20:54.25*** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell)
20:54.25*** mode/#asterisk [+o Qwell] by ChanServ
20:54.25husimonzap show channels?
20:54.37husimondrmessano, slipstreaming is your friend
20:54.44drmessano-LTI know
20:54.54*** join/#asterisk Wangster (n=Wangster@host-253.epicnet.ca)
20:54.55drmessano-LTBut my slipstream disk is very custm
20:54.57drmessano-LTcustom
20:54.58husimonas well as never using windows again
20:55.05husimonthat's also your friend
20:55.11drmessano-LTand I needed a more generic install for this machine
20:55.13WangsterWhy the heck does my asterisk 1.4 keep over-writing my voicemail.conf file??! How do i stop it?
20:55.24husimonhttp://goodbye-microsoft.com/
20:55.34husimonthat's my favorites site to tell peole to goto :)
20:55.36drmessano-LTWell, we can argue that crap all day long.. you wanna port these apps to Linux? lol
20:55.44husimondrmessano, what apps?
20:56.04drmessano-LTCustom stuff we use at work... So drop the rhetoric.. I get it :)
20:56.28husimondrmessano, yeah I'm not completely anti windows, I know sometimes you just have to use it.
20:56.29*** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net)
20:56.33husimondrmessano, I like to avoid it is all
20:56.38drmessano-LTDitto
20:56.44drmessano-LTBut sometimes you just can't
20:56.55husimondrmessano, when I want to crank out some games, windows xp is on my dual boot.
20:56.59husimon;)
20:57.06*** part/#asterisk gdiebel (n=gregd@adsl-69-217-146-185.dsl.mdsnwi.ameritech.net)
20:57.18husimonin os x I have parallels with windows, that's just about the perfect marriage of two os'
20:57.22drmessano-LTI was thinking this morning how much I would prefer this running on Linux
20:57.30drmessano-LTBut :(
20:57.33husimondrmessano, you could always try wine
20:57.41*** part/#asterisk RoyKa (n=roy@ip-77-15-149-91.dialup.ice.no)
20:57.49husimonit might be worth a few minutes of trying it
20:57.58drmessano-LTNot worth the hassle for a single tasked machine
20:58.10drmessano-LTMay as well install XP and forget it
20:58.38jameswfI got Photoshop running in wine... almost cried..
20:59.00husimonjameswf, hehe
20:59.18*** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr)
20:59.19drmessano-LTBesides, as with a lot of specialized apps, this one would end up with some croak in wine anyway
20:59.24drmessano-LTThats always the luck
20:59.31husimonyeah
20:59.36husimonyou are not bad off with windows xp
20:59.41husimonwindows vista can suck my ballzzzz
20:59.42drmessano-LTor my 4 port serial card would burp
20:59.45drmessano-LTor some crap
20:59.45jameswfdrmessano-LT: vmware..........
21:00.01husimonjameswf, yeah, but you still gotta run windows :)
21:00.09husimondoesn't help you not install all the patches and sps
21:00.12jameswfi have like 8 installs from the comfort of my desktop
21:00.13drmessano-LTYeah, still not worth it.. this machine is getting one app and then being shoved in a closet
21:00.27husimonmore complexity to break
21:01.01jameswfthe last windows box I touched ran windows 3.1
21:01.11jameswfof course that was this morning
21:01.13pkunkradoes anyone use teliax here?
21:01.13husimonjameswf, hehe that was my first windows box
21:01.16drmessano-LTI could always throw AsteriskWin on there lol
21:01.22husimoncourse before that the box had dos
21:01.28husimonasteriskwin?
21:01.31pkunkrai use them right now but i'm getting dropped calls.
21:01.33drmessano-LTDon't ask
21:01.47jameswfI sent an email to asteriskwin32 still havent heard back
21:01.50pkunkratrying to figure out if the issue is on their end or mine.
21:01.59drmessano-LTjameswf: three months ago? lol
21:02.01husimonhow can I use asterisk to listen in on calls in the network?
21:02.09husimonlike dial a special number and bridge into a call muted.
21:02.15drmessano-LTAsk AT&T
21:02.22jameswf~chanspy
21:02.22jbotchanspy is probably an application that adds the ability to spy on any bridged call, this includes VoIP only calls where ZapScan/ZapBarge couldn't this can. As of october 19 2004, ChanSpy is not included in the standard Asterisk distribution or the development CVS tree.
21:02.37*** join/#asterisk cpjosh (n=josh@cp120.cardplayer.com)
21:02.38J4k3~at&t
21:02.38jbotit has been said that at&t is the devil.
21:02.40J4k3~att
21:02.45husimonso much power... :)
21:02.52jameswf~devil
21:02.53jbotThey say if you play a Windows CD backwards you hear satanic messages. What's even more scary is if you play it forward, it installs Windows.
21:03.06husimonit's neat to watch a convo in ztmonitor
21:03.07pkunkrahahah
21:03.09husimonwatching it go back and forth
21:03.19drmessano-LTJesus.. I am NEVER going to use the browser on this box.. WHY make me upgrade to IE7
21:03.28drmessano-LTDamn it all to hell!
21:03.31husimondrmessano, you can remove it
21:03.32denonnothing's making you upgrade
21:03.36denonjust click No
21:03.39denonand it won't ask again
21:03.39drmessano-LTI know
21:03.40husimondrmessano, i remove ie7 on all my xp boxes
21:04.20drmessano-LTActually, the date of forced upgrade is coming
21:04.25drmessano-LTMar 1 maybe
21:04.55drmessano-LTNo
21:04.57drmessano-LTFeb 12th
21:04.58*** join/#asterisk bkruse (n=bkruse@216.207.245.1)
21:04.58*** mode/#asterisk [+o bkruse] by ChanServ
21:05.22*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
21:05.36pkunkrai believe windows update is making you use ie7
21:05.41*** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
21:05.41J4k3whats scary about microsoft is every time you upgrade you end up with MORE bugs
21:05.45J4k3correct one, add five.
21:05.58cpjoshCan anyone give me some advice on this? I have a working asterisk install and can dial in and out, however when dialing out on a second line i get a 'line is busy' and the following error in the log: app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown)
21:06.00[TK]D-FenderJ4k3: Just like COBOL
21:06.11cpjoshI have a zaptel trunk setup and it has 24 channels...
21:06.25pkunkraie6 has a whole slew of its own bugs.
21:06.27drmessano-LTJ4k3, thats interesting math... how do you explain the 117 to 1 ratio with Vista? lol
21:07.10J4k3drmessano-LT: I explain vista like this...   I sit there for a few minutes, quietly, looking good as I always do... then I crap my pants violently and fall over and start convulsing
21:07.20husimoncpjosh, pastebin your extensions.conf
21:07.20drmessano-LTHA
21:07.23J4k3then I make sure to take as much of everyone's time as possible in the process of cleaning myself back up
21:07.29husimoncpjosh, are you using a group for your trunk, or a single channel
21:07.52hmmhesayshmm do dynamic features work in meetme?
21:08.03husimoncpjosh, for instance on my outgoing patterns I use Zap/g1
21:08.05drmessano-LT"I'm really not into Pokemon"
21:09.21cpjoshhusimon: my outbound pattern is ZAP/1
21:09.35drmessano-LTWow
21:09.42husimoncpjosh, I don't know much about this but I'd say try ZAP/g1
21:09.48drmessano-LTSo the Dell Trixbox Pro system won't be named Trixbox afterall
21:10.00J4k3Dell DudeBox
21:10.08drmessano-LT"Fonality VoIP Phone System"
21:10.18J4k3where all your calls get sent through to the Indian Voice Translation Service
21:10.19cpjoshhusimon: oh you think im referencing a channel instead of a channel group?
21:10.23husimoncpjosh,  yes
21:10.27jameswf~fonality
21:10.30cpjoshhusimon: Ah!
21:10.32J4k3so your organization can go from professional to twinkie-salesman overnight.
21:10.33drmessano-LTThat's damn interesting
21:10.35cpjoshhusimon: thank you very much.
21:10.49drmessano-LTI was wondering if Dell was going to REALLY market something with a name like Trixbox
21:10.53denonI do not know dell dudebox, please kindly hold while I look up your question in our kindly knowedgebase articles sir
21:10.54drmessano-LTI guess I know the answer
21:11.10*** part/#asterisk Synoptic (i=Synoptic@modemcable034.152-81-70.mc.videotron.ca)
21:11.15jameswf~fonality is <reply> Fonality is hiring no Asterisk knowledge needed or, I just installed asterisk now what?
21:11.17jbotokay, jameswf
21:11.17Qwelldrmessano: Fonality is calling it "foncore"
21:11.34J4k3drmessano-LT: Dell has cornered the market with a whole line of PCs that anyone with a clue refers to as "shitbox"
21:11.40*** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr)
21:11.49husimonJ4k3, true but they are cheap
21:11.51jameswf~dell
21:11.52jbotDude! Are you getting a Dell?, or stupid for only selling PCs with Vista, or might be cooler if they start pre-packaging OpenOffice per the OpenOffice.org request sent to the CEO
21:11.54husimonJ4k3, throw a 3 year on them and who cares
21:11.59J4k3husimon: crap hardware is never cheap
21:12.17J4k3downtime > warranty
21:12.18husimonJ4k3, I agree for certain applications they are a bad idea
21:12.25denonJ4k3: funny you should say that, knowing our involvement in openwrt
21:12.34drmessano-LTQwell: Is foncore the name of the butchered Asterisk, or the whole technology?
21:12.37b11d.
21:12.39husimonJ4k3, but for peoples workstations and general purpose crap machines they are fine, especially if you keep on in reserve and image your installs
21:12.42J4k3I mean, if you're buying an office load of crap for useless employees... buy them some old 1st gen imacs used and save yourself a lot of money
21:13.10husimonkeep one in reserver*
21:13.14husimonreserve.
21:13.24davenor don't and spare yourself the headache of repairing the drives in those fuckers
21:13.26J4k3denon: thats the part that kaloz pissed me off the other day.  I've ran openwrt on a LOT of hardware, and madwifi on even more hardware than that.
21:13.49husimonI have to say dells case design has gotten a lot better.
21:13.55J4k3denon: thats when I concluded that openwrt was never going anywhere and left.
21:14.07J4k3husimon: considering a cardboard box was better than their old cases....
21:14.12husimoni hate clamshell
21:14.16husimonis what I mean to say
21:14.18J4k3I hate tool-less.
21:14.23husimonwhy?
21:14.27drmessano-LT"Dude, I got a PBX"
21:14.32husimonlol drmessano
21:14.33J4k3because it generally causes two things
21:14.35J4k3A> idiots in PCs
21:14.41J4k3B> having to beat up the box to get in the SOB
21:14.56husimonJ4k3, I guess I don't have that problem because my users don't open the boxes.
21:15.00J4k3dell's newish microatx chassis requires a beating to get inside
21:15.03trixbox-fanboyI need help
21:15.07drmessano-LTThats NOT far from SOME of the Trixbox users.. I wont say all, or most, but some are just.. WINDOZ4LIFE
21:15.17husimonJ4k3, I have to admit once, with an ibm case I couldnt' figure out how to get it open :P
21:15.24husimonJ4k3, fucking secret release lever
21:15.25drmessano-LTyes, trixbox-fanboy, wut can I helpz you wit?
21:15.28J4k3heh drmessano-LT, you've heard about the put-asterisk-on-win32 project right?
21:15.40drmessano-LTI've seen AsteriskWin32
21:15.49husimontrixbox-fanboy, windows and trixbox and asterisk oh my!
21:15.50trixbox-fanboyI ran yum update now zaptel is broked
21:15.57J4k3omfg, windows can't even bring up the start menu in a reasonable period of time... how the hell is it going to perform in a network situation with devices that notice a 20ms jitter.
21:16.12drmessano-LTfoncore will fix All that
21:16.21drmessano-LTfoncore will save the world
21:16.42husimonfoncore vs chuck norris
21:16.53husimonfirst round chuck will be in here crying
21:16.55drmessano-LTDellnality PhonesHome PBXcellent Phone System will rock balls
21:17.21drmessano-LTI CAN HAZ AKERISK?
21:17.38outtoluncno its MINE
21:17.40outtoluncall mine
21:17.42J4k3GrandDell DellTone 101
21:17.48drmessano-LTOh god
21:17.49husimonYou remember that old t-shirt, reading your emails for fun and profit?
21:17.53drmessano-LTyes
21:17.58husimonI want an asterisk t-shirt that says "listening into your calls for fun and profit"
21:18.22drmessano-LT"You know that thing your wife does?  I heard you and her talking about it."
21:18.28husimonand on the front have a man entry for chanspy
21:18.34husimonor the back
21:18.47husimonLAUGH
21:18.51husimonbig white letters
21:18.54drmessano-LTlol
21:18.56husimon<front> CHANSPY
21:19.03husimon<back> Listening to your calls for fun and profit
21:19.04husimonlmao
21:19.07J4k3ASTERISK CAT LISTENS WHILE YOU PHONESEXOR
21:19.15drmessano-LTI would NOT trust a Fonality system, and here Dell is selling them.. WTF..
21:19.17husimonthat's such a perfect shirt for asterisk geeks
21:19.20drmessano-LTDoes that mean I can be rich too?
21:19.30J4k3drmessano-LT: maybe you shouldn't trust dell?
21:19.35drmessano-LTASTERISK CAT <----- HAHAHAHAH
21:19.48husimoni don't get the cat bit?
21:19.57drmessano-LTGoogle for "ceiling cat"
21:20.09husimonah
21:20.19J4k3http://rudd-o.com/wp-content/uploads/images/funny/Ceiling_cat_is_watching_you_masturbate.jpg
21:20.55husimonpeople have a retarded fascination with cats
21:20.58husimonlolcats ...
21:21.05J4k3lolicats
21:21.05drmessano-LTLOL
21:21.06[TK]D-Fenderheading home, BBIAB
21:21.09J4k3pre-teen pussy
21:21.33*** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner)
21:21.33husimonJ4k3, your preference?
21:21.41J4k3husimon: nah, my cats are ooooold
21:21.42drmessano-LTAsterisk cat is in ur foncore, stealing your SIPs
21:21.44jameswf-homeirc doesnt like the nick changes
21:21.56tobiascan folks here recommend a particular VOIP provider for a small asterisk setup?
21:22.08husimon"I'm in your asterisk stealing your sips
21:22.10husimon"
21:22.20tobiasI've been using VoicePulse but a network issue between our server and theirs has made it practically unusable
21:22.35husimonhmm
21:22.39husimoni need to make that asterisk shirt
21:22.44husimonwhere can I make custom t-shirts online
21:22.55Qwellhusimon: cafepress
21:23.02J4k3tobias: I love that.  my itsp changed internet connections and things weren't right for 2 months after
21:23.10QwellI assume they're still around
21:24.36drmessano-LTthey are
21:25.02husimonwhat the crap
21:25.05husimoni can't customize the back of the shirt
21:25.12Qwelltry harder
21:25.24husimonoh I will
21:25.37tobiasJ4k3: yeah.  i think it was working ok for awhile.  traceroutes to voicepulse look great from my other servers across the country too.
21:25.56jameswfis asterisk compattible with web 2.0
21:26.14tobiasthough really, voicepulse should have servers in more than one location.
21:26.17Corydon76-vcchI dunno, are you compatible with blue cars?
21:26.46*** join/#asterisk servergod (n=servergo@70.97.159.120)
21:27.05tobiasCorydon76-vcch: your question still makes more sense than his :p
21:28.19servergodhi all
21:28.52husimonQwell, nope can't figure out a way to get stuff on the back
21:28.55J4k3tobias: yeah, I suspect the problem is all these itsp's are too small potatos to do it right
21:29.13maszloi can not figure out this caller id problem at all.  i have debugged from asterisk, viewed the full log and i can not see where this callerid is coming from.  it shows the correct callerid, until it hits my cell phone, then it changes
21:29.27J4k3you've got the big companies that won't speak to you with less than 1M/minutes/month or more, then you've got everyone else who's scratching over that 8th of a cent per minute profit they're making.
21:29.36husimonit appears you can't print on the back of black shirts
21:29.42husimonhow stupid is that
21:29.54J4k3asterisk crosses me as more of a blue shirt kind of subject.
21:30.17husimonchanspy doesnt
21:30.47maszlodoes anyone have experience with running asterisk on an verizon pri?
21:31.02servergodyes
21:31.13maszlocan you set the callerid?
21:31.25*** part/#asterisk javar (n=javar@69.79.134.24)
21:31.32mvanbaak"I drive a blue Toyota Prius"
21:31.41jameswfgayyyyyyyyyyyyyyyyyyyy
21:31.51servergodas long as it matches a DID in their CNAM database
21:32.00J4k3"I ran over a blue toyota prius in the driveway, oops"
21:32.01mvanbaak"do you know what sound that makes when you drive by ?"
21:32.10mvanbaak"Iiiiiiiiiiiiiiiiiiiiiiiiiiiiiii'm gay"
21:32.15J4k3I keep a thick layer of mud on the bottom of my truck so I can drive over economy cars and not scratch the paint.
21:32.16maszlodo i have to enter the number without a name?
21:32.40jameswfI took my wife to go see duhnum live funny stuff
21:33.11mvanbaakSILENCE! I KIIIIIL YOU "
21:33.29*** join/#asterisk ShaunWing (n=chatzill@dsl-243-73-60.telkomadsl.co.za)
21:33.36servergodVerizon uses the DID on the sip:nxxnxxxxx@foo.com to do an auth and a lookup. no custom id is accepted or used, since they do a dip on the CNAM database
21:33.43servergodop
21:33.45servergodpri
21:33.45ShaunWingHelp asterisk -r does not let me reconnect
21:33.55servergodsorry, pri
21:34.07mvanbaakShaunWing: ps ax | grep asterisk
21:34.29ShaunWing<PROTECTED>
21:34.31ShaunWing<PROTECTED>
21:34.40servergodIf you want a custom name on a DID, you have to submit it to verizon
21:35.11maszloi just want it to get correct number on outbound calls
21:35.26mvanbaakShaunWing: what's the error
21:35.43servergodon a specific trunk? or just for specific extensions?
21:35.56maszlohave been playing with this all day with no luck, "sip:nxxnxxxxx@foo.com" can you explain this
21:36.07maszloall line show the same outbound callerid
21:36.10servergodsorry, that is for sip not a pri
21:36.23maszlowhat is a automated system, not even a calling number
21:36.27*** join/#asterisk UnixDog (n=unixdog@ppp-71-128-114-104.dsl.irvnca.pacbell.net)
21:36.30servergodusing freepbx or just cle?
21:36.37servergod*cli
21:36.41maszlofreepbx
21:36.44ShaunWing[root@localhost asterisk]# asterisk -r
21:36.45Qwell#freepbx
21:36.45ShaunWingAsterisk 1.4.14, Copyright (C) 1999 - 2007 Digium, Inc. and others.
21:36.47ShaunWingcertain conditions. Type 'core show license' for details.
21:36.48ShaunWing=========================================================================
21:36.50ShaunWingHANGS HERE
21:37.11*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:37.12mvanbaakShaunWing: and asterisk is working ok ?
21:37.16mvanbaakor not working at all
21:37.18ShaunWingyes
21:37.35*** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com)
21:37.39ShaunWingI'm happy to just kill it but how and then restart it
21:37.40servergodedit the trunk, and under general outbound caller id
21:37.47ShaunWingDon't want to reboot my server
21:37.56*** part/#asterisk UnixDog (n=unixdog@ppp-71-128-114-104.dsl.irvnca.pacbell.net)
21:38.19servergodShaunWing
21:38.31ShaunWingyes
21:38.34mvanbaakShaunWing: killall -9 asterisk
21:39.08jameswfkillall -9 silly_beavers
21:39.16maszlomy current outbound callerid is blank in the trunk
21:39.58servergodset it there, and do the reload and call, see if the CID chages
21:40.09ShaunWingTx that worked
21:40.20maszlowont that make it for all lines on that trunk?
21:40.54servergodon a specific trunk? or just for specific extensions? That is why i asked that
21:40.55Qwell~freepbx
21:40.56jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
21:41.16drmessano-LTOh god
21:41.23drmessano-LT"Hacking Trixbox"
21:41.27drmessano-LTI can just see it now
21:41.30mvanbaak~closedpbx
21:41.33maszloi have 12 extensions on one trunk, if i set the callerid on the trunk wont they all have that as the callerid?
21:41.39servergodhow can i share a trunk on *1.4 if i send a sip call to it from ccm?
21:41.41drmessano-LT"I hacked my trixbox and removed all the GUI stuff.. how cool is that?"
21:41.42Qwellmaszlo: there is no such thing as a trunk
21:42.01kyrondrmessano-LT, maaan...you're like so 1337
21:42.13maszloQwell: what do you mean?
21:42.17mvanbaakdrmessano-LT: svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk
21:42.18mvanbaak;)
21:42.26drmessano-LT"Yeah, I stripped so much out of it, it comes up in freakin DOS"
21:42.32*** join/#asterisk riksta (n=rick@rhamnett.plus.com)
21:42.37Qwell~siptrunk
21:42.37jbotThere is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example.  You set up a SIP trunk like a regular peer in sip.conf.
21:42.47Qwellmaszlo: go to #freepbx, we cannot help you with that here
21:43.22drmessano-LTAll bashing aside
21:43.23mvanbaakwe should update jbot
21:43.24kyrondrmessano-LT, I have to use a _keyboard_ and there is NOTHING to click!!!....talk about retarded!
21:43.31rikstaHi, I'm originating a channel and I want to record separate CDRs for both legs, for billing purposes. I'm using the ForkCDR command but it does not work as expected, the fork makes the first leg's bill seconds 0. What is the proper way to log both legs correctly?
21:43.32kyronpun
21:43.44mvanbaakfreepbx is a virus that will render you asterisk unusable in most cases
21:43.46drmessano-LTWhy would you want to install a trixbox system, and THEN use it to experiment with
21:44.11*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
21:44.11*** mode/#asterisk [+o lmadsen] by ChanServ
21:44.19lmadsenANSWEREDTIME is in seconds I presume.. ?
21:44.19mvanbaaklmadsen !!!!!!!!!!!1
21:44.25lmadsenmvanbaak: !!!!!!1one!
21:44.32mvanbaaklol
21:44.44Qwelldrmessano: speaking of book titles...  I love what Kerry named his book.
21:44.44mvanbaakthey still did not merge it :(
21:44.54Qwell"trixbox made easy"...as if it were ever difficult
21:44.55lmadsenthose bastards!
21:45.01drmessano-LTYeah, no crap
21:45.08mvanbaaklmadsen: I can merge it.......
21:45.15lmadsenmvanbaak: mwahahaha
21:45.45*** join/#asterisk Robba (n=rob@203.56.181.15)
21:45.50mvanbaakwould be funny, my 3rd commit to -trunk being 50K of changes
21:45.51drmessano-LTWell, it takes a real tool to take something as easy as Trixbox, make it out to be hard, and then write a 500 page book on it
21:46.00*** join/#asterisk CrashSys (n=kumba@t1.databalance.com)
21:46.21drmessano-LTIf I wrote a book on trixbox, I could read it in one sitting on the shitter
21:46.25plikthere's another book called "trixbox without tears"... which seems unlikely to me
21:46.41*** join/#asterisk atis_work (n=atis_wor@81.198.164.2)
21:46.44Robbai'd have to agree with that
21:46.57drmessano-LTMy CentOS + Asterisk + FreePBX guide is called "Asterisk without Tricks" lol
21:47.04plikha
21:47.05ShaunWingBy the way.. Anyone registered Asterisk to Net2Phone?
21:47.08Robbalol
21:47.49drmessano-LTI was damn proud of that name
21:47.56mvanbaakasterisk without tears => vim /etc/asterisk/*
21:48.13*** join/#asterisk chavigny (n=nrp@c-67-171-147-26.hsd1.or.comcast.net)
21:49.37chavignyAnyone have and sugguestions, Im looking for a carrier that can sell me some minutes and a few dids in an areacode of my choice/plus toll free, i looked at gzlink but its not that good
21:50.02plik~itsp-us
21:50.18drmessano-LT168 pages..  Chapter 1: Click
21:50.19plik~itsplist-us
21:50.19jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com
21:50.26drmessano-LTChapter 2: Unclick
21:51.00*** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl)
21:51.02roxluhi there
21:51.09Robbaanyone have any idea as to why i can't use (CallerID(num)=number)
21:51.15roxluis it possible to see what 'calls' are currently being made?
21:51.16mvanbaakChapter 3: get lost in the spaghetti code in your dialplan
21:51.33Robbait just keeps saying module not registered
21:51.52Robbabut if i run  module load func_callerid.so
21:52.06mvanbaakRobba: pastebin the dialplan lines
21:52.14*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:52.18mvanbaak~pb
21:52.19jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
21:52.24mvanbaak~tk
21:52.25jbotACTION snipes $herlo with a straw and rolled up piece of paper
21:53.06mvanbaakyou all have to start worry now, [TK]D-Fender is here
21:53.16mvanbaak;)
21:53.19[TK]D-Fender:O
21:53.44mvanbaak=4
21:54.09Robbahttp://rafb.net/p/aLzd9M84.html
21:54.10ShaunWingSay Any idea why there is no speech in either direction when use Quintum registered to Asterisk. Quintum on its own works fine
21:56.03mvanbaakRobba: what version of asterisk
21:56.10Robba1.4.17
21:57.21mvanbaakRobba: try in all caps
21:57.30Robbaok
21:57.42*** join/#asterisk djweis (n=djweis@67.55.197.226)
21:57.56mvanbaakexten => s,1,Set(CALLERID(num)=0398982277)
21:57.58ShaunWingSay no one heard of Quintum?
21:59.04Robbaok i did that now its calling out from a random number from our 100 number range
21:59.07[TK]D-FenderRobba, and like I told you before "_s" is NOT a pattern.  it must be just "s"
21:59.17[TK]D-FenderRobba, Go fix your contexts
21:59.42[TK]D-FenderRobba, 58 Polo
22:00.00*** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble)
22:00.02[TK]D-FenderRobba, is this the same broken thing I gave you the play-by-play on already?
22:00.08Robbano
22:00.30Robbai have changed from setcallerid to callerid(num)=
22:01.32[TK]D-FenderRobba, you still seem to have a lot of the stuff I commented on already unfixed.  Also you have "congestion" following a Macro call you never come back from.  wastage.
22:01.43*** part/#asterisk cpjosh (n=josh@cp120.cardplayer.com)
22:03.14mvanbaakwaste -> wastega
22:03.14Robbaso i shouldn't have those lines at all?
22:03.25mvanbaakfinal fantasy ftw !
22:05.48[TK]D-FenderRobba, Clearly not
22:07.41J4k3hmm does voicepulse filter callerid?
22:08.02J4k3er toward
22:08.47[TK]D-FenderJ4k3, Not to my awareness
22:09.05twistedoh holy crap
22:09.14twistedsvnfs <3
22:09.41Robbahttp://rafb.net/p/SXgGdz57.html
22:10.42twistedyou know you have too much stuff open when it takes 5 minutes to get to the point the system will restart
22:11.13[TK]D-FenderRobba, [awc-phones] and [incoming] are very redundant and you may have missed 1 entry
22:12.49Robbai used [incoming] because that was the context asterisk was looking for, for the extension 7600
22:12.53[TK]D-FenderRobba, exten => 101,1,VoiceMailMain(${CALLERID}@${CONTEXT}) <-- after everything we went through, STILL not right
22:13.27*** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com)
22:13.59*** part/#asterisk maszlo (n=reckenro@65.223.240.146)
22:14.44Robba[TK] i'm still not sure what that is supposed to be
22:14.50*** join/#asterisk fujin (n=aj@unaffiliated/fujin)
22:16.23husimon[TK]D-Fender, do you have any idea what would cause intermittant echo?  Do you think the hardware echo cancellation is trying to cancel but fails once in a while?
22:16.54husimon[TK]D-Fender, if you remember from yesterday i'm running a phonebridge2 with hardware ec.
22:17.00[TK]D-FenderRobba, "core show function CALLERID" and go reread chapter 5.
22:17.15[TK]D-Fenderhusimon, yes, entirely possible that it just sucks
22:17.21husimon[TK]D-Fender, i'm thinking that too
22:17.34[TK]D-Fenderhusimon, Don't say you weren't warned
22:17.37husimon[TK]D-Fender, but then again, the number I had a problem with has a really long run between the phone and the pbx
22:17.44husimon[TK]D-Fender, we have historically had problems with that office
22:18.15*** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted)
22:18.15*** mode/#asterisk [+o twisted] by ChanServ
22:18.16twistedoops.
22:18.21twistedhelps not to kill the term window :P
22:18.32husimon[TK]D-Fender, the asterisk box and this hardware is a huge improvement over the old setup.  I can actually use the speaker phones now, before calling that office would completely echo out and breakup while using a speaker phone.  But other lines were fine.
22:23.37*** part/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl)
22:23.39husimonhehe i made the t-shirt http://kolea.ifa.hawaii.edu/~nhuisman/front.jpg http://kolea.ifa.hawaii.edu/~nhuisman/back.jpg
22:25.11twistedmight i suggest getting digium's permission to use the asterisk logo?
22:25.19husimontwisted, yeah i'm not actually making it...
22:25.20husimonheh
22:25.24twistedaww
22:25.26husimonprobably just do Asterisk *
22:25.31husimonor a big *
22:26.15husimontwisted, it's pretty far on the geek scale of things :P
22:26.33[TK]D-Fenderor a bull's eye with a hole punched out & fake blood-stain around it
22:26.50husimonhehe
22:27.12*** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net)
22:27.26pliktshirts must be a hot topic just now... this just got posted in another channel http://www.themishmash.com/2008/02/10-actual-t-shi.html
22:27.26[TK]D-Fenderslightly off-center so as to look plausible
22:27.31x86I've got an old legacy Toshiba phone system that I want to connect with Asterisk over a T1 interface
22:27.35lmadsengrrr... I'm getting errors when I place a call that my ODBC connection is down, but 1) it isn't, and 2) I'm setup to write CDR records to the DB.... so I've no idea why this is happening... *sight*
22:27.38lmadsen-t
22:27.41x86the Toshiba is setup already using the T1 interface to the PSTN
22:27.51x86it's using E&M Wink to the PSTN
22:27.54husimonplig tshirthell.com
22:27.57husimonplik
22:28.11Juggielmadsen, cdr_odbc or cdr_adaptive_odbc?
22:28.17lmadsenJuggie: shouldn't be...
22:28.20x86I was wondering if I could just set E&M Wink on the interface on the Asterisk side, and it would work correctly?
22:28.20plikbeen there, done that... so have those guys
22:28.26x86do I need a T1 crossover cable or something?
22:28.31lmadsenJuggie: maybe just having the module loaded does it though
22:28.41lmadsenlet me try that
22:28.57Juggielmadsen, do you want to write cdr's w/ odbc?
22:29.03lmadsenJuggie: no, I don't
22:29.13lmadsennot yet anyways... but I didn't configure it to write them
22:29.18Juggiethen noload both those modules.
22:29.30Juggiesome old config you have setup is making one of them load
22:30.38lmadsenJuggie: same BS
22:30.45lmadsenJuggie: this is a new config
22:30.48husimonwow I can't find any asterisk t-shirts
22:31.03*** join/#asterisk angryuser (i=nononon@df01t2-212-194-235-109.d4.club-internet.fr)
22:31.08lmadsenit should not be trying to connect to the DB....
22:31.14lmadsenthere's no reason for it to
22:31.19husimonbeside of course from digium
22:31.27Juggielmadsen, 'module show like cdr'
22:31.28husimonwith their butt ugly orange shirt
22:31.29lmadsenoh well, will have to fix this later... conference call with another client
22:31.42Juggiesee which of them is loaded (via use count)
22:31.50drmessano-LTI have an Asterisk shirt ;)
22:31.55lmadsencdr_manager.so                 Asterisk Manager Interface CDR Backend   0
22:31.55lmadsenfunc_cdr.so                    CDR dialplan function                    0
22:31.55lmadsencdr_csv.so                     Comma Separated Values CDR Backend       0
22:31.55lmadsencdr_custom.so                  Customizable Comma Separated Values CDR  0
22:31.59husimondrmessano, did you check out the one I made?
22:32.04drmessano-LTNo
22:32.08husimonhttp://kolea.ifa.hawaii.edu/~nhuisman/front.jpg http://kolea.ifa.hawaii.edu/~nhuisman/back.jpg
22:32.28drmessano-LTHAHAHHAHAHHAH
22:32.38husimon:)
22:32.47Juggielmadsen, paste bin the errors?
22:32.57*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
22:33.03husimoni'm super tempted to cafepress that
22:33.10drmessano-LTMy day is done.. be back on from home later
22:33.15husimonwith a non-copyright digum logo.
22:33.18lmadsenbasically does this... but there is no reason to be connecting to the database for any reason at this point....
22:33.19lmadsen[Feb  7 17:30:23] WARNING[9023]: res_odbc.c:103 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [FreeTDS][SQL Server]Error converting data type varchar to bigint. (66)
22:33.19lmadsen[Feb  7 17:30:23] WARNING[9023]: res_odbc.c:111 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect...
22:33.27*** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au)
22:33.40lmadsencdr_odbc.conf is commented out
22:33.55lmadsen1.4, so no cdr_adaptive_odbc to evenbe loaded
22:34.31Juggielmadsen, cdr_odbc doesnt use res_odbc anyways i dont think
22:34.36Juggieunless its been upgraded recentally.
22:35.16*** join/#asterisk codejunky (n=jan@codejunky.org)
22:35.31Juggieweird.... i would have to see the box to know more but odd.
22:35.43lmadsenya... very odd...
22:35.46lmadsenprobably a bug...
22:35.55lmadsenbecause there really is no reason to be trying to write a CDR to the DB
22:35.56husimonwhat happens with hardware and software echo cancellation are used together?  FUBAR?
22:35.58Juggiepossible
22:36.05Juggiebecause when a module refuses to load
22:36.08Juggieit just refuses
22:36.11Juggiebut it doesnt really not load
22:36.21lmadsenthe funniest thing is I'm trying to make something NOT work :)
22:36.37Juggiedid you do a noload on cdr_adaptive_odbc and restart?
22:36.45husimonin modules.conf
22:36.45lmadsencdr_adaptive_odbc is not in 1.4
22:37.02lmadsenbut I added it to the modules.conf in a noload anyways
22:37.07lmadsenand I did 'restart now'
22:37.10*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)
22:38.10Juggielmadsen, that is odd, because i dont think cdr_odbc uses res_odbc
22:38.24lmadsenya.... then this is even more confusing
22:38.45Juggiedoing an update of my 1.4 code now to look
22:38.51lmadsenDial(SIP/${EXTEN}@${PROVIDER}) should nto cause a DB connection at all
22:39.07lmadsendoes it after the Dial(), and after the Hangup()
22:39.30Juggieya, in 1.4 cdr_odbc definitally does not use res_odbc so far as i can tell
22:39.37Juggieit would have a call to SQLPrepareAndExecute
22:39.40Juggiebut does not.
22:40.14lirakislater all
22:40.21*** part/#asterisk lirakis (i=lirakis@66.252.24.133)
22:40.34*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
22:40.53*** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net)
22:41.03CrazyTuxHey guys, how do I allow a bigger inbox
22:41.09CrazyTuxFor voicemail?
22:41.26Juggielmadsen, ya, odd, no call. has to be something else it cant be cdr_odbc
22:41.34lmadsenya...
22:41.50Juggiegood luck, i'm gone :)
22:41.52lmadsenlates
22:42.09Juggieodbc voicemail?
22:42.21CrazyTuxJuggie, yea mysql
22:42.32JuggieCrazyTux, was talking to lmadsen.
22:42.37CrazyTuxJuggie, ah
22:42.47Juggielmadsen, try disabeling the freetds connection, which ever module is using it should complani.
22:42.49lmadsenJuggie: ya... but no call to Voicemail()
22:42.52Juggie*complain.
22:42.58lmadsenI'll figure something out
22:43.04Juggielmadsen, doesnt matter.
22:43.08lmadsenlates
22:43.16Juggieit would still get called every minute to check mwi,
22:43.18grandpapadotHello all.  On Aastra 480i CT phones, if I have 3 lines provisioned, will they all use g729a (if so configured in the sip peer) or will only the first line use g729a and the rest fallback to ulaw (like some other phones and/or ata's?)
22:44.04lmadsengrandpapadot: asterisk would have no concept of the separate lines if they are all the same peer configuration
22:44.30CrazyTuxDoes anyone here know if I can increase the size of a specific mailbox in asterisk?
22:44.34grandpapadot@lmadsen: 3 differnet peers.  My question was really on the phones codec implementation/support, will it do 3 simultaneous g729a calls.
22:45.17grandpapadotCrazyTux: I had an email here someone that was saying how to increase the size.. oh, wait, that was for my penis, nevermind.
22:46.32*** join/#asterisk AndyGraybeal (n=andy@node191.34.251.72.1dial.com)
22:46.32*** join/#asterisk servergod (n=servergo@70.97.159.120)
22:47.06CrazyTuxgrandpapadot, haha.
22:47.28lmadsengrandpapadot: not sure... depends on the phone I guess
22:49.09servergodhave 1 *1.4 and a cisco ccm.  have sip trunk from asterisk to voip provider. CCM uses pri. want to send LD from ccm to asterisk.  have trunk from ccm to * when call ld i get * saying that extension dont exist. How can i route it out?
22:50.03husimonwhat the hell, if I make a constant sound in my phone the other persons voice cuts out
22:50.25grandpapadothusimon: The nerve!
22:50.37husimonno say i call an IVR somewhere
22:50.41husimonthen hum a note
22:50.48husimontheir voice all but cuts out and i get bits and pieces
22:50.57husimonwonder what is causing that
22:51.11grandpapadothusimon: Yea, I hear ya. More information might be helpful to get a good answer from someone.
22:51.25husimonnot sure what information to give
22:51.35grandpapadothusimon: What phone?  How is asterisk configured?  Analog, SIP, PRI, etc?
22:52.04husimoncisco 7940, normal asterisk configuration, no echo cancelation or tx/rx adjustments, sip, over a pri.
22:52.21husimongrandpapadot, are you on an asterisk system right now?
22:52.28husimoni'm curious, call 1800polycom
22:52.29grandpapadotYes, 1.2.26
22:52.30husimonand try humming
22:52.37husimonlike sing a note and see if you can hear the prompter
22:52.47ManxPowerTurn off VAD/CNG on the phone
22:52.58husimonManxPower, what are those?
22:53.01grandpapadotlol
22:53.08ManxPowerIf you can only hear the far end when you make noise -- then it's prolly a VAd/CNG issue
22:53.20husimonManxPower, it's not that I can only hear them when I make noise
22:53.24grandpapadotManxPower: Call 1-800-Polycom, guy's trolling
22:53.26husimonit's that when I make a constant noise they cut out
22:53.36husimonpfft screw that it's just the first number I thought of
22:53.49ManxPowerhusimon: those are options you should disable on the phone.  I don't know how or where those options are, as I understand it Cisco phones do NOT default to VAD/CNG.
22:53.54grandpapadotIt's a sex line...
22:54.08grandpapadotfunny, though, that 1-800-polycom is a sex line
22:54.11ManxPowerOh, so when YOU send something, the far end cuts out?
22:54.23husimonManxPower, yeah not normal conversation
22:54.28husimonbut if I hum a constant note
22:54.50husimontheir side cuts in and out, now I haven't tried this with a real person yet just noticed it while trying to test for echo
22:54.50grandpapadothusimon: That's just noise cancellation on the phone probably.
22:54.55husimongrandpapadot, ah
22:57.20beightoI am having a serious problem with an Asterisk system I installed.  It seems that about 50% of the non-local calls (800 or long distance) don't go through.  There are 3 pots lines hooked up and when dialed out upon say "you must dial a 1 before completing this call" from the local phone company.  This happens at random and not on any 1 particular line out of the 3.  I have it dialing a 1 in...
22:57.21beighto...the dialplan and just to make sure I even added a + to the front with no change.  I switched from inband to RFC2833 with no change.  I cranked up the tx gain to 10 with no change.  Any ideas?
22:58.43husimonbeighto, sounds like your dialplan is wrong
22:58.54husimonbeighto, or one of you 3 lines is different then the other two
22:59.17beightohusimon: I have checked it over and over.  The console shows the 1 being dialed and all 3 pots lines do the same thing at random.
22:59.29husimonbeighto, random for the same number
22:59.38beightohusimon: yes
22:59.53husimonbeighto, you should probably pastebin your extensions.conf
22:59.56husimonso we can look at it
23:00.05beightookay, just a minute
23:00.44husimonalso a sanity check would be to plug a phone into that analog line and make the same calls and see what happens
23:02.59J4k3report: low
23:03.07husimonhehe
23:03.25J4k3what I'm getting tired of is my damned cellphone
23:03.46J4k3call my *, dtmf works fine for the ivr... get to a human and the echo canceler goes 180 degrees out of whack
23:03.50J4k3they can BARELY hear me
23:04.05J4k3but if I beat on the dtmf a bit, it'll work again
23:04.06husimoni'm probably just going to leave the users to deal with the small bit of echo we have
23:04.10husimonit's only to one office
23:04.19husimonwhere their stupid loop is wayyyyy too long
23:04.28J4k3but I suspect its my cellphone...  the other cellphone here (same provider/technology) doesn't do it
23:04.28husimonlocal loop i mean
23:05.00J4k3husimon: you could always get the telco to install some PAIRGAIN hardware
23:05.02J4k3*screams*
23:05.04beightohusimon: http://pastebin.com/d489db2ee
23:05.28ManxPowerhusimon: HPEC is your friend if you do not have many channels and a fast system
23:05.50*** join/#asterisk PepOSX (n=angeldav@190.72.132.46)
23:06.10husimonManxPower, hpec is software or is that the ec built into digiums card?
23:06.43*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
23:06.47husimonlaugh they sell it for $10, why not just make it free
23:07.01*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
23:07.01*** mode/#asterisk [+o anthm] by ChanServ
23:07.05grandpapadotUsing static realtime in 1.2.2x, what will initiate a musiconhold class reload that's stored in the db?  A reload doesn't seem to do it.
23:07.13grandpapadotRestarting asterisk does, however.
23:08.26grandpapadotAlso, is there a way to 'list' loaded musiconhold classes from the cli?
23:08.29*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:08.35husimonbeighto, so with that config you dial a given number, it works. Then dial it again and it doesn't work?
23:08.46beightohusimon: correct
23:09.02*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
23:09.34grandpapadotugh, nevermind, lol
23:09.34pkunkraalright, i'm trying to figure something out.  my callers call my pbx located in my datacenter as normal, the go into the queue and it rings my extension, as soon as i pick up, it disconnects.
23:09.37husimonwhat happens if you remove one of your patterns and only provide the _1N.....
23:10.04husimonpkunkra, i'd go into the cli and set verbose to 9 and watch
23:10.05pkunkrathere are no errors in the logs anywhere.
23:10.12pkunkraok
23:11.07husimondoes anyone know how I can get a telco test number to adjust my tx and rx gain?
23:11.09beightohusimon:  I would image that wouldn't make any difference.  I only put the _NXX pattern there just in case they are too lazy to dial the 1.
23:11.19pkunkraproblem is, it is intermittent.
23:11.21husimonbeighto, shrug it's worth a quick try.
23:11.33husimonpkunkra, keep trying until it happens.
23:11.49husimonpkunkra, use your cell
23:12.30pkunkraalright.  i'll give it a try
23:12.32husimonbeighto, Try plugging in a real analog phone to the first line and see if it's fine.
23:12.47beightohusimon: Yes, analog phones work fine
23:13.02beightohusimon:  I even had a different phone company run 4 new lines
23:13.16beighto3 new lines 1 was for faxing
23:13.32husimonyou've tested all  right?
23:13.39husimonall three
23:13.41husimoni mean
23:13.49beightoyes, I plugged a fax machine into all the lines and tested them
23:14.08beightothat was my version of an analog phone at the time
23:14.20husimonI mean plug in the analog phone and dial a number you know has problems and make sure each line works with it.
23:15.02beightoyes
23:15.32beightodid that the first day
23:15.37husimonI might suggest putting some NOOP statements in there to echo ${EXTEN}
23:15.46husimonto see what your * is actually getting
23:15.54beightohusimon:  I am using presence too, but I wouldn't imagine that would effect it either
23:16.25beightothe SIP debug shows the 1 being dialed as well...
23:17.04beightoI haven't done a NOOP, but the dial statement clearly shows the whole number
23:17.13husimontrue
23:17.23beightoIt is driving me crazy!
23:17.47husimonbeighto, i'm fairly new to *  so that's about all I got.
23:17.56beightobummer
23:18.11husimonbeighto, you should ask [TK]D-Fender sometime, make sure to provide lots of logs.  He is pretty much the local expert.
23:18.28beightoI am wondering if maybe the polarity is reversed on the lines, but I think that shows up on the console
23:19.28beighto[TK]D-Fender:  Are you there?  Care to chime in?
23:19.42*** join/#asterisk pkunkra (n=chris@cpe-72-229-148-29.nyc.res.rr.com)
23:19.51pkunkraargh.
23:19.55phix[TK]D-Fender: hey, regarding echos and feedback from ATA, I still experience it, the changes made to gain settings have only seem to come into effect when going through the landline, not SIP.
23:20.14beightohusimon: Thanks for trying
23:20.15husimonphix you might be able to adjust the tx and rx for the phones
23:20.18husimonnp
23:20.50ManxPowerphix: that would be because you can't have echo on a all VoIP call (at least traditional echo, you can still get feedback between the mic/speaker that would be echoy"
23:21.08*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
23:21.27ManxPowerAnd in fact, you can't have loss on an all VoIP call, so there is no reason to change the audio gain
23:22.39husimonManxPower, so where exactly does HPEC sit?  inside asterisk? or outside or?
23:23.06ManxPowerhusimon: It's software.
23:23.09Qwellhusimon: in zaptel
23:23.13husimonah
23:23.20husimondoes it require digium hardware?
23:23.24Qwellnope
23:23.31husimoncause I have tdmoe
23:23.48ManxPowerit DOES require zaptel.  I doubt it will do you any good with TDMoE
23:23.58ManxPowerYou might have to talk to Digium
23:24.05husimonyeah probably doubtful
23:24.14husimoni do have a digium card I could test it with
23:25.00ManxPowerYou could use a Sangoma or other zaptel compat boards as well
23:25.42husimondoes anyone know where I can read how to use the milliwatt application to do tx rx testing?
23:26.00Qwellshow application milliwatt?
23:26.42husimonyeah i see how to use it in a dialplan, just wondering if there were any tricks
23:26.50lmadsenummm.... it plays a tone...
23:26.53lmadsennot much else to figure out
23:26.58*** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk)
23:27.26husimonyeah so then what do you adjust your tx/rx to,
23:27.46husimonthe wiki says 100%
23:28.06husimonfor the phone telco test number
23:28.09ManxPowerthe wiki is wrong half the time
23:28.14ManxPoweryou need ztmonitor I believe
23:28.32husimoni'm not even sure using milliwatt on the local side gives me anything
23:29.16pkunkraok.  so i've called my phone about ten times now.  can't reproduce.  but it seems to only happen with calls from other parts of the U.S.
23:29.42husimonpkunkra, i guess the best I can say is next time it happens go back into your cli and find that time and call
23:29.51husimonpkunkra, might want to output your cli to a text log file
23:31.05pkunkrahusimon, now, here's the catch,  a customer called me and i had the problem. then i called with my call, no issue.  i then had the customer call back and it was the same issue.  this all happened in about five minutes.
23:31.20pkunkrahusiman, how do i output it to a text log?
23:31.26pkunkrathat'd be useful.
23:31.30husimonpkunkra, i'm looking that up now
23:32.16pkunkraok
23:33.01ManxPower/etc/asterisk/logger.conf
23:35.38husimonManxPower, i see that it lets you specify levels of logging to different files
23:35.49husimonbut how do you actually get the cli output in a log?
23:37.49pkunkrai suppose i could stuff it inside a screen or a script
23:38.18husimonit seems like you should be able to output it a log file
23:39.25pkunkrai would think so.
23:39.29pkunkrathat's an important feature
23:39.32husimoneither i'm missing something
23:39.39husimonor ...
23:39.56*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
23:40.54pkunkranone of the options in logger.conf seem to be a reproduction of the cli output.
23:41.06pkunkrait does dump some stuff in /var/log/asterisk/*
23:41.14pkunkrabut it isn't all that useful
23:42.19husimonanyone care to comment?
23:43.16pkunkraok.  i have a hunch.
23:43.20pkunkralet me try something.
23:43.39pkunkrai'm going to start up a slew of traffic
23:43.39pkunkrabittorrent, etc.
23:43.48pkunkraand see if that might make the problem reappear again.
23:44.27pkunkraproblem seems to happen in the middle of the day.  busiest times, like 2-3pm.
23:46.29husimonpkunkra, oh you have a sip trunk?
23:47.23*** join/#asterisk craigk (n=craigk@58.174.150.119)
23:48.22pkunkrait is sip.  not sure if its a trunk or not.
23:49.15*** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com)
23:49.31pkunkragoogled.
23:49.46pkunkrait does run over the same wires as the regular internet traffic.
23:49.59pkunkrai just did a test call.  lots of jitter and delay but no dropping.
23:49.59Qwell~siptrunk
23:50.00jbotThere is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example.  You set up a SIP trunk like a regular peer in sip.conf.
23:50.11pkunkrahah
23:50.26pkunkraits a regular sip peer then.
23:50.26Qwellmeh, somebody changed that
23:50.55pkunkra( i did have an IAX2 trunk before.  it sucked and i switched it to sip )
23:51.16*** part/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net)
23:51.24pkunkraseems sip is much better at handling jittery networks than iax2 is
23:52.19pkunkrabut i do use an external provider to terminate the phone lines.  the do a DID into my pbx in the datacenter.
23:52.27pkunkrathe pbx then calls my extension.
23:52.37husimonobviously there must be some distinction between a sip peer that is a phone and that is a provider, so what do you use in conversation when you are talking about a sip peer that is your provider? "sip provider" ?
23:53.13jameswfhusimon: the distinction is context
23:53.16husimonobviously you are trunking your traffic across that peer
23:53.36husimonseems like you should be able to use the word trunk.
23:53.39*** join/#asterisk Ad-Hoc (n=nimbus@88.218.87.95)
23:54.05jameswfI got junk in my trunk hey
23:54.08husimonor is it just nope, use peer always and be smart enough to figure it out.
23:54.12pkunkrahah
23:55.12husimonjameswf, do you know of a way to log the messages on the cli to a file?  the logger.conf doesn't really seem to do it. unless i'm totally stupid and don't understand the logger.conf (which is highly possible)
23:55.38jameswfyes add verbose
23:56.18husimonso   "file = options,options,verbose"
23:56.18pkunkrai'm thinking the issue could be with one of two points in the network.  either my cable router in my home office.  or the provider i buy the voip from
23:56.23husimonerr =>
23:57.14husimonk jameswf thanks
23:57.17husimonpkunkra, you catch that?
23:57.27pkunkrai think i might be able to figure it out
23:57.33pkunkralooking in logger.conf now
23:57.53*** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net)
23:58.13husimonbtw i'm sure glad that was documented....
23:58.29mchouanyone here happen to use linksys wrtp54g?
23:58.44mchouI mean the ATA portion
23:58.47husimongonna add that to the wiki
23:58.49husimonverbose = cli....

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