IRC log for #asterisk on 20080301

00:01.53coolhpIs anyone having problems getting the blindxfer feature to work ? I have "Tt" passed as options in my Dial() command and DTMF is set to RFC2833 on both sides... automon works if I set DYNAMIC_FEATURES but blindxfer never does :-(
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00:14.02LemensTS.
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00:20.05actros1840hi
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00:22.16HyphenexWhat is AMP and would I have it if I built asterisk from source?
00:22.41ManxPower~amp
00:22.41jbotamp is, like, NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
00:23.29ManxPowercoolhp: try "Ww
00:27.33Hyphenexokies, any guides on getting asterisk to work with FWD without AMP then?
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00:37.26ManxPowerAT&T sent me an e-mail advertizing a new product.  It was so filled with crappy HTML that my html cleaning script basically deleted the entire contents of the message.
00:37.45ManxPowerHyphenex: Um, have you tried the FWD site or the Wiki?
00:38.07ManxPowerAll AMP is, is a GUI for pussies that don't want to edit config files.
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00:38.41kiscokidI have a Polycom phone question
00:39.40ManxPowerkiscokid: ask it
00:39.41kiscokidAnyone know what makes the call timer start on a Polycom?  Sometimes it doesn't start right away.
00:40.04ManxPowerkiscokid: no idea, I would imagine it would be trigered by the far end answering
00:40.11kiscokidright
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00:40.39kiscokidbut sometimes when I call a 800 number the signal is not received for a long time
00:40.43actros1840nobody use MP3Player ?
00:41.16sbingnerkiscokid, I've called 1800 numbers where they don't actually answer till I choose the first option
00:41.24kiscokidseems like if the signal is not sent by the remote end within 60 secs the phone decides to hang up
00:41.26sbingnerI think they're cheating on long distance charges that way
00:41.44HyphenexLittle question.  Can a user belong to more then one contexts?
00:42.32kiscokidDoes the Polycom or * have some parameter that I could change to make it wait for 120 secs?
00:44.01kiscokidalso, any know the name for that signal?
00:44.59kiscokidsbinger: the particular 800 number I am having trouble with doesn't send the signal until way into the call menu
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01:05.18stansmith0
01:06.22ManxPowerkiscokid: put an Answer before the Dial
01:08.33ManxPowerkiscokid: I suggest you read thru the Admin Guide to see if anything looks useful.
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01:14.29ManxPowerkiscokid: You are NOT the only one that has calls that ring for more than 60 seconds
01:16.06stansmithlol
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01:25.06kamitodoJust got a second-hand SPA2102.  My setup is DSL modem -> OpenWrt -> SPA2102 plus all other LAN and WIFI devices.  Cannot access the SPA2102 via the browser.  Any ideas?
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01:26.52lyroyDoes someone here have experience on how to setup a provisioning server for linksys phone adapter like pap2 and wrt54gp2 ?
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01:33.58kiscokidManxPower: ?
01:35.20kiscokidManxPower: should I put Answer before Dial for every call?
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01:38.57stansmithlol?
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01:49.00ManxPowerkiscokid: that is up to you.
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01:55.24kamitodoIs anyone familiar with SPA2102?
01:56.23kiscokidManxpower: I don't understand the implications of your answers.
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02:02.27MatBoyit's really weird that when I use in a2billing my cardnumber and my password, I never can authenticate
02:02.34MatBoyalways password wrong in the errorlog
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02:10.46mchouanyone here have experience with linksys wrtp54g (wireless router with 2 voice ports)?
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02:11.36coppiceI have one
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02:12.12mchoucoppice: what ver firmware you have loaded on it?
02:12.27MatBoymhh, I don't get it, maybe I need sleep :)
02:12.39coppicesomething old. they never seem to update it. it sucks
02:12.39MatBoybut this is actually quite easy to setup with a2billing
02:13.01mchoucoppice: you need frequent resets for that device?
02:13.27mchoucoppice: like it "crashes" on you?
02:13.40MatBoywhen you can login to a2billing as a user, can you also authenticate with the same details ?
02:13.53coppiceyeah. everyone complains about the same thing. if you never make a call it seems to work Ok as a wireless router, it seems to be telephony that makes it fall over
02:14.14mchoucoppice: sigh.
02:14.43mchoucoppice: is pap2 more relaible or just as bad?
02:14.52mchoureliable*
02:15.07coppicei don't hear complaints about the PAP2 crashing
02:15.41mchoucoppice: crap.  maybe I should have bought pap2 instead.
02:15.51MatBoyyeah !!
02:15.53MatBoydid it :)
02:16.17MatBoyguys I love you :P
02:16.38TJNIImchou: I have a couple PAP2s.  No complaints.  All eBay specials.
02:17.24mchouTJNII: what's the going price for those babies?
02:18.12TJNIIDepends on demand.  The last ones I got were $30ea, but I paid too much.
02:18.21mchouand is the dlink VTA the same thing?  cause if so I'd like to just go to Fry's to pick them up
02:18.47coppicemchou: I'd advise against getting babies. they are even more troublesome than those linksys boxes
02:19.09TJNIIDon't know.  You just have to watch for firmware locks when you bou off the shelf.  If you're willing to hack, more power to you.
02:19.49mchouwell, apparently the dlink vtas are harder to hack for whatever reason
02:20.14mchouevent though some have said the HW is exactly the same
02:20.27mchouas a pap2, that is
02:21.51mchouvoip HW is costing me a mint, testing junk that's not reliable :(
02:22.19mchouhad such high expectations for wrtp54g
02:23.29TJNIIHeh.  I've had good luck with the pap2s.  I've got a bunch of budgetone-100s that are good if you overlook the poor caller ID and speakerphone.
02:24.02TJNIISome buddies of mine have some nice polycom-500s, but they manage to find a mislabled eBay auction to get them cheap.
02:24.43mchouI dont think I'm ready for a full blown IP phone yet at this stage
02:24.55mchoustill prefer ATAs
02:25.16TJNIII got a SNOM 220 yesterday, I have high hopes for that.
02:25.17mchoujust cause voip HW is such a crapshoot
02:25.44TJNIIHeh.  Doing an ebay search for sip phone hasn't led me wrong yet.
02:25.50mchouTJNII: new or used?
02:26.03TJNIIWell, except that IP0021 phone.  That thing is junk.
02:26.24coppiceall current VoIP hardware is crappy. its just a question of how crappy. one or two phones are starting to look like their maker has a clue, though
02:26.38TJNIINew
02:27.20TJNIIThe SNOM core is just a powerPC chip running linux.  I probably won't be able to resist hacking it.
02:27.47mchoucoppice: which phone makers have a clue?
02:27.59coppicethe power PC seems to be growing in these little embedded boxes
02:28.42coppicewell, in some markets polycom seems to sell decent phone for <$100. most people say nice things about the value for money of those
02:29.18TJNIIThe Polycom 500s look nice, usually go for around $100 on eBay
02:29.27_ShrikEIMHO polycom's low end phones have been crap until the 320/330's came out.
02:29.54coppiceyeah, those 320s seem to be designed to shake up the market
02:30.49TJNIII say the budgetones make good starter phones.  Not expensive, good call quality on the handset, easy to set up.
02:31.04TJNIIThough I know "budgetone" is a naughty word here.......
02:31.08kiscokidGrandstream?
02:31.11TJNIIyea
02:31.16kamitodoanyone familiar with SPA2102?  I got version firmware version 5.1.12 and don't know the default admin pass.
02:31.21mchouTJNII: so what exactly is wrong with caller ID on budgetone?
02:31.27TJNIINumeric only
02:31.35TJNIINo letters, just the number
02:31.43coppiceI think if grandstream just got a competant designer to do a nicer case for them, people would have a far higher opinion of them
02:31.53TJNIIHeh
02:32.14TJNIIMine look kinda like hotel phones.  They look fine in the bedroom, not so much on my desk.
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02:33.34mchoukamitodo: is it a locked version?
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02:34.45kamitodomchou: I don't know.  Bought a used one off amazon.  How do I tell?
02:34.53mchoulol
02:35.02mchouif it's used it's locked :)
02:35.29mchoukamitodo: you better read up on how to hack it
02:35.30kamitodowell, is it for sure?  how do i check?
02:35.53kamitodohm...  any experience with that?
02:36.14mchoukamitodo: plsa do your own homework on that
02:36.18mchoupls*
02:36.57mchouplenty of sites give general ways to unlock the locked devices.
02:37.00mchouYMMV
02:37.35kamitodowell, OK.  what if it is not locked?  what's the default pass then?
02:39.12mchoukamitodo: should be in the linksys documentation
02:39.39mchoutry admin/admin
02:42.20kamitodoadmin/admin does't work.
02:42.37kamitodoi see on the box hat the model no. is spa2102-na
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03:08.22TJNIIGAAH!
03:08.41TJNIINever was good with 'dem things.
03:09.03obnauticusIS anyone here good with a2billing?
03:09.32kamitodothere *must* be a way to tell whether the damn thing is locked!  referring to the spa2102 of course.
03:09.36obnauticusbecause for some reason when i add a prefix as a rate, a2billing is still not forwarding the call.
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03:15.15WilliamKkamitodo, tried the manual for it yet?
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04:07.12outtolunc..
04:07.24drmessano..
04:07.54drmessano@?
04:08.07outtoluncas to the lack of scrolling *not* happening here
04:08.52drmessanoah
04:09.07drmessanoThings have been quieter
04:09.53coppice......zzzzzzZZZZZZZZ
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04:25.37LoofAnyone here familiar with comdial PBX systems... it seems that I can put a T1 interface card on an asterisk box and basically have talk T1 interface to T1 interface as a sort of pass-through
04:25.52LoofAt least, that's how I'm understanding the description I'm seeing
04:25.57Loofam I offbase here?
04:27.38Loofhmm
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04:48.16x86what's comdial?
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04:58.59rfernandezhi! a p 4 with 3.0 ghz and 1gb in ram may handle 24 trunks with 24 extensions?
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05:28.44obnauticushow do i pass inline operators for agi scripts from the dialplan?
05:29.04obnauticuslike say i have script.agi then ./script.agi x y z
05:30.19obnauticusand i want to pipe stdout from that script into Festival, so how do I do that?
05:31.36obnauticuswow
05:32.04obnauticusthat crapped on my question
05:32.04outtoluncbummer
05:32.17obnauticusdo you know
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05:34.59outtolunci'd have to scroll back
05:35.24outtoluncuse |
05:36.02outtoluncexten => _X.,5,AGI(dnidlookup.agi|${CALLERID(num)},${IF(${ISNULL(${CALLERID(dnid)})}?${EXTEN}:${CALLERID(dnid)})})
05:36.06outtoluncas an example
05:37.09outtoluncanother example
05:37.17outtoluncexten => 7,n,AGI(disporec.agi|${OPTION},"399",${CAMPAIGN},${LEADID},"10")
05:37.27obnauticusouttolunc like say i want a user to be able to enter a number
05:37.30obnauticusinto their keypad
05:37.33obnauticusi do a waitexten
05:37.41obnauticusand you use that number as an in-line operator
05:37.43obnauticushow do i do that
05:37.46obnauticusor an argument
05:37.56outtoluncyou have to use read for that
05:38.20outtoluncor background
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05:38.57obnauticusthen how do i pipe the stdout from the script
05:38.57obnauticusto agi
05:38.57obnauticusthen how do i pipe the stdout from the script to agi
05:39.00outtoluncyou don't you read it to a var, then use the ${var} when you get back into dialplan
05:39.47obnauticus>: |
05:39.47obnauticuswhat do you mean?
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05:40.26outtolunchttp://dynx.net/ASTERISK/gnudialer/agiIVR.agi
05:40.26outtoluncread that
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05:40.28obnauticusk
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05:40.28outtoluncit give *alot* of examples
05:40.28outtoluncgives even
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05:41.04outtoluncsearch for 'question_1' that directly relates to your question
05:41.04obnauticusthanks
05:41.31obnauticusso like
05:41.37obnauticusthe read sets the DTMF tones to variables
05:41.44obnauticusread from the user's channel?
05:42.04outtoluncyes
05:42.10outtoluncplay a prompt
05:42.20outtolunc*read* input
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05:42.27outtoluncstore to var
05:42.29obnauticushttp://www.voip-info.org/wiki/view/Asterisk+cmd+Read
05:42.32obnauticusim reading that right now
05:42.35obnauticusand your other script
05:44.44obnauticusouttolunc so how do i store stdout to a variable?
05:45.38outtoluncyou mean stdin
05:45.45obnauticusno stdout from the agi script
05:45.50outtoluncyou don't
05:45.53obnauticus:\
05:45.58obnauticusI have an NPA-NXX lookup thing im trying to make..
05:46.07outtoluncyou store stdin, to the var as use it as ${VAR}
05:47.49outtolunchttp://dynx.net/ASTERISK/AGI/agi-ccard.agi
05:48.05obnauticushttp://www.monetra.com/~brad/callerid_shell.agi
05:48.10obnauticusim actually trying to get that to work
05:48.11obnauticus:\
05:48.26*** join/#asterisk horseradish (i=user@63-76-119-176.directcom.com)
05:49.19horseradishi need advice on an asterisk setup w/ equipment
05:49.45outtolunclook at the last line
05:50.19outtolunconce that agi returns to the dialplan you use it as ${lookupname}
05:50.39obnauticuswell see i want the user to input their own number.
05:50.43obnauticusbecause right now it just gets it from the callerid
05:50.56obnauticuscaller id number*
05:51.03obnauticusi.e. ; exten => s,1,AGI(callerid_shell.agi|${CALLERIDNUM})
05:51.05horseradishi want a 2 line SOHO setup. with 2 voicemail boxes, an introductory message when ppl call, ability to place people on hold, send them to voicemail boxes, transfer calls, and an after-hours message
05:51.05outtoluncyes, you supply a NUMBER, it finds a NAME
05:51.18obnauticusya im making my * server to read for a number
05:51.30horseradishi'm not sure if it would be better to buy my own box and run it on a POTS line or if i should go with a VoIP provider
05:52.25outtolunchorseradish: note that using the available hardware on standard 1mb's *usually* does not give you things like disconnect supervision
05:53.14obnauticusouttolunc oh i know what you mean
05:53.26outtoluncbingo!
05:53.27obnauticusit then lets you use it later as $lookupname
05:53.28outtolunchehe
05:53.32obnauticuserr
05:53.33obnauticuscool
05:53.38outtoluncas ${lookupname}
05:53.43obnauticusand that would be
05:53.57outtoluncexten => s,1,AGI(callerid_shell.agi|${CALLERIDNUM})
05:54.02obnauticusexten => s,1,Festival(${lookupname})
05:54.16obnauticushttp://pastebin.ca/924299
05:54.16obnauticusthat
05:54.18obnauticuss what i haev written
05:54.19outtoluncexten => s,2,NoOp(callerid name is ${lookupname})
05:54.45outtoluncyou can't have 2 s,1's
05:54.47obnauticusand the script can read the number as NPA-NXX-XXXX
05:54.56obnauticusi know that i was going to edit the priorities later
05:54.57obnauticuslol
05:55.15outtoluncyou got 3
05:55.25obnauticusbecause i knew it was going to be priority 3
05:55.36obnauticusi write it, then i do priorities... it's my thing
05:55.37obnauticuslol
05:55.49outtoluncextension processing on a 'reload' stops at the first error
05:56.01outtolunc(which you should see with verbose 3 or higher on the cli)
05:56.03obnauticusi would have noticed
05:56.08obnauticusya
05:56.14obnauticusim just an AGI nub
05:56.14obnauticuslol
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05:56.38outtoluncjust use s,1   then the rest use s,n
05:56.51obnauticusheh
05:57.50obnauticuserrrr
06:00.47obnauticusouttolunc how do i make it so it also echo's the Company name
06:01.22obnauticuswait i think it already does for ${company}
06:01.41outtoluncif the script returns that also, store it to a variable also (look at the last line)
06:02.39obnauticusi don't think it returns that
06:02.39obnauticus:"\
06:03.34outtoluncit doesn't 'myname' gets replaced along the way with whatever 'name ish' thing it finds
06:04.42obnauticusSo :\
06:06.14obnauticusoh outtolunc you can change the "lookup order"
06:06.18obnauticusso it finds 411 info too
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06:06.36outtoluncyes, i get that <G>
06:07.17horseradishis vonage a good voip provider for asterisk stuff
06:07.30obnauticusno, use voipjet
06:07.33horseradishor what is a good voip provider for my 2 line soho
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06:11.09LoofSo, can a T1 interface talk to another?  I.e. something that expects to hook up to the c/o T1 talking to a T1 interface on an Asterisk box?
06:11.28jqlLoof: yes
06:12.05horseradishif i pay for a voip provider, then asterisk providers all the features like call waiting, call queing, hold music, etc. right?
06:12.09jqlideally, you're using pri, which means you set the asterisk box to transmit pri-net
06:12.15horseradishand i can have unlimited lines and phone numbers if i go voip, right?
06:12.34horseradishcan i get a 1-800 number with someone like voipjet
06:12.44scooby2anyway to flap or loop a t1 on a sangoma card?
06:12.45LoofOk, that makes sense of what I'm seeing in the FAQs then
06:13.42LoofSo if we've got a dual T1 PBX and want to keep our existing system and use DID for call routing... we need a quad T1 card on the asterisk box
06:13.58Loof2 in from c/o and 2 out to the old PBX
06:14.35TJNIIhorseradish: Well, not unlimited.  There are bandwidth restrictions and restriction from your sip provider.
06:14.41TJNIIAnd yes you cen get a 1800
06:14.46jqlLoof: that would make sense, yes
06:15.21jqlLoof: I'm sure the FAQs and manuals all mention it, but make sure your quad card gets its clock from the c/o
06:15.31TJNIIhmmm.... fwd and iax just doesn't wanna go......
06:15.43LoofI've been trying to find a good FAQ or manual or howto for it
06:16.02LoofI've found forum answers that -seemed- to indicate that
06:16.06Loofbut I wasn't sure if that work :P :)
06:16.15jqlI'll confirm it. :)
06:16.28LoofDo we need a csu/dsu between the PBX and the asterisk box?
06:16.38LoofI'd assume not, from what I see of the ports.
06:17.03LoofAnd I would guess we need a straight wire cable, not a cross over
06:17.09jqlno, just needs cat5 cable
06:17.32jqlT1 cross-over is different than ethernet crossover
06:17.35jqland you do need it
06:17.42jqlget a wiring diagram
06:19.08LoofDuh, found it :)
06:19.09drmessanoIIRC you swap orange and blue
06:19.22LoofShould've googled for t1 crossover in the 1st place :)
06:19.26jqlheh
06:19.55LoofThe only thing I'm not sure of... what happens when someone hits a transfer button or vm button within the pbx system
06:20.13LoofDoes that pass back up the chain in some way?
06:20.26jqlnot likely
06:20.33Loofi.e. will we be able to assign extensions to and transfer between voip phones and the pbx phones?
06:20.37jqlthe phone company doesn't care about your voicemail
06:21.11jqlyour voip phones can certainly dial the pbx with enough motivation
06:21.47jqlthe pbx may require voodoo to know to dial out for what look like internal extensions
06:21.58jqlthey all have the voodoo, but it's always voodoo
06:22.08jqlprepare a virgin for sacrifice
06:22.31LoofOk, so from the perspective of asterisk these look like a whole bunch of POTS interface phones... and any voip phones are what they are.
06:22.57jqlpretty much
06:23.28LoofThus we'd have some kind of DTMF interface to the functions... #<something>,<ext> to transfer while on a call or something
06:23.47jqlquite practical of you
06:23.56LoofSorry, not a telecom guy :)
06:23.59LoofIf its not %100 obvious
06:24.09LoofI'm on the IT side of the universe with a little telecom experience
06:24.17Loofenough to hurt my head provisioning a T1 now and then
06:25.03jqlt1 here, and a pbx there, and soon you're talking business
06:25.04LoofAnd I think virgin sacrifice is against company policy... we can request chicken entrails with forms in triplicate
06:25.07Loof;)
06:25.42jqlentrails entreat the spirits, but virgins appease the Gods
06:25.43LoofOk, so from the perspective of the end users... if we do this right... they will end up with a few dozen 'useless' buttons on their phones.
06:26.07LoofSomething that will actually work transfering pbx<->pbx... but fail pbx<->sip device.
06:26.22LoofThus, we'll retrain to use the DTMF codes for transfer, vm access, etc.
06:26.36jqldepends on how opaque the pbx configuration is
06:26.59jqlsometimes they have a mapping of extension 4xxx => dial 7005554xxx
06:27.01LoofUnless the phones can be reprogrammed to do some sort of signaling via DTMF for the call features
06:27.06jqlwhich would go to asterisk
06:27.07Loofbut that seems like too much to ask for
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06:28.43Loofit's a comdial DXP
06:29.16LoofSo, I think that type of mapping should work... but call transfer and VM are internal, as far as I gather
06:32.46jqlold school pbxs are old school
06:33.44jqlthat comdial has an 1100 page manual
06:34.59LoofI wonder if it has some way to do in-band signaling then...
06:35.00Loofhmm
06:35.41LoofI assume there is no other way to salvage the phone and wiring infrastructure without continuing to use the PBX controller
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06:37.09jqlif there is, the PBX installer was a fool
06:37.16jqlerr... I mean... perhaps. :)
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06:38.28jqlwell hell
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06:38.42jqlyou can indeed reprogram all the buttons on the phone using this disgusting manual. :)
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06:48.08Star568please teach me how to autostart asterisk on centos version 4
06:49.25BeeBuuStar568: run asterisk
06:49.42*** part/#asterisk horseradish (i=user@63-76-119-176.directcom.com)
06:51.28Star568i want asterisk can auto start the service after system reboot
06:54.30BeeBuuadd it in /etc/init.d/
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07:03.53Star568BeeBuu: i am a newbie to linux, where and how can i find the startup script for asterisk, just cp the file to this directory?
07:04.12_mwoodj_Star568: 'make config' will install the scripts. 'chkconfig asterisk on' will enable it for centos
07:05.49_mwoodj_See the "Installing Asterisk" section of this document: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
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07:06.48Star568thanks all for the information :)
07:06.53_mwoodj_no problem
07:17.03Loofjql: Thanks for the help, that was exactly the boost I needed to figure out the rest.
07:18.44Star568_mwoodj_: it works now :D
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09:05.41BeeBuu~book
09:05.41jbotAsterisk: 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, or see ~buybook
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09:31.12BeeBuuanyone still here now?
09:31.20obnauticus/me
09:31.44BeeBuucan i dial on console?
09:31.51obnauticusvia console?
09:31.55BeeBuuya
09:32.05obnauticusyou can use err
09:32.08obnauticusOriginate.
09:32.17obnauticusorigionate with the application dial
09:32.43BeeBuucan i dial with Priority under console?
09:33.40BeeBuuhere is what i want: make the * system auto call some people and play a message
09:34.08BeeBuuor meet them
09:34.09obnauticusyou can make an agi script to run in the background
09:34.23*** join/#asterisk PepOSX (n=angeldav@190.72.146.136)
09:34.26BeeBuua morning meet... :-)
09:34.42obnauticusthat osunds wrong
09:34.55BeeBuuobnauticus: what's wrong?
09:35.02obnauticuserg
09:35.14BeeBuuerg?
09:35.21obnauticuserg.
09:35.52BeeBuuwhat's wrong?
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11:25.58_gmanyone here tries realtime ldap driver?
11:26.01_gmtried*
11:39.56^shark_hi guys i am trying to use the cfgfmt.exe tool to change the .txt file to binary for the 7912g phone but i dont understand what -FLAG mean in here >> cfgfmt.exe -FLAG sip_ptag.dat gkdefault.cfg.txt gkdefault.cfg, is it an option of its self? i certainly dont see it reflected here >> http://cisco.com/en/US/docs/voice_ip_comm/cuipph/7905g_7912g/3_3/h323/english/administration/guide/7905conf.html#wp1052210
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11:50.49^shark_is this (-FLAG) supposed to mean an  RC4Password ? or something else?
11:52.38^shark_or nothing at all >> does this make more sense if i am not using  RC4Password -- cfgfmt.exe -sip_ptag.dat gkdefault.cfg.txt gkdefault.cfg
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12:02.00^shark_i am getting error >> error: can't open ptag file: 'ptag.dat' who can kindly help me with this?
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12:06.19^shark_any body atleast read my query?
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12:36.15ice_croft!book
12:36.23ice_croft~book
12:36.24jbotAsterisk: 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, or see ~buybook
12:37.16ice_crofthi all
12:39.59ice_croftwhere can i read some about * interconnection?
12:40.09ice_croft*'s
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13:03.45ectoice_croft:  what do you mean?  IAX connections, SIP connections, etc?
13:04.00ectoSetting up an IAX connection between two Asterisk boxes is in TFOT book
13:04.04riddleboxcan anyone help me with sla? I have configured it according to doc/sla.pdf and when I do sla show trunks and sla show stations there is nothing, http://pastebin.ca/924527, there is nothing
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13:15.44ice_croftecto> i found it already, thanx
13:23.20^shark_hi
13:24.42^shark_i am trying to boot the 7912g ip phone, and i have my dhcp/tftp server (192.168.100.166) and the files that the phone is looking for which you can see from this >>  http://internetworkpro.org/pastebin/2013
13:24.44*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
13:25.05^shark_but still the phone will not boot, any ideas why this is happening?
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14:05.09dacsgood morning every one
14:07.53*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:08.19dacsi have been asked by our church priest, to find a telephone device that can store 100+ phone numbers so that he can record a message regarding service time and date and send it at once to all of the numbers.! is that possible to do with *?
14:10.36_ShrikEdacs: yes, it's possible
14:11.11riddleboxdacs, astercrm
14:12.11dacsriddlebox: where can i get more info please
14:12.23riddleboxwww.astercrm.org
14:12.53riddleboxdacs, the developer is working on getting it to work with fxo trunks now, may be a couple of weeks
14:12.59riddleboxbut it works with sip trunks now
14:13.01dacsalso what is confusing me , how is that possible :) , shouldn't the phone ring first to go to vmail
14:13.36riddleboxso the priest wants to call vmail boxes? or call peoples houses?
14:14.13dacsriddlebox: people houses /cell . because right now he dails every single phone,
14:14.32riddleboxasterisk+astercrm, will be able to handle it
14:15.09dacsriddlebox: or maybe be like IVR , call all those numbers at once and play the msg to who ever picks
14:15.09_ShrikEdacs: If he wants to send the 100 calls all at once, you will need to get him a voip provider or about 4 T1s
14:15.11riddleboxwhen solo, gets the fxo trunks to do it
14:15.26riddleboxyou can set it and forget it!
14:16.16riddleboxdacs, my boss has me creating an asterisk box with 4 fxo ports in order to that very thing
14:16.17dacsriddlebox: i like the solo, since expness will come out of my poket, i hate it when priest do that
14:16.49riddleboxwell how much are yo willing to spend?
14:16.54dacsthey tell you we need this , this and that and GOD bless you
14:17.00riddleboxhey _ShrikE  have you ever dealt with sla?
14:17.46dacsriddlebox: there is an old box in the church i will check it and see if i can load linux and then *
14:17.47_ShrikEI try to avoid it :)
14:18.26riddleboxdacs, good luck
14:18.48riddlebox_ShrikE, I have a customer who really wants it and doesnt like idea of not seeing which lines are in use
14:20.56dacsriddlebox: do you mind if you help me please
14:21.12_ShrikEriddlebox: If they just want to see what lines are in use, then hints and BLF would work.
14:21.21riddleboxdacs, I can try
14:21.41riddlebox_ShrikE, they want to press it and dial and put people on hold
14:22.28dacsriddlebox: thank you so much, so when i meet with the priest today i will explain to him what i am thinking to do , and how long this project will take!
14:22.52dacsriddlebox: can you check your pm
14:23.02riddleboxdacs, it may be two weeks or so for the developer  to get it working right
14:23.44riddlebox_ShrikE, did you see my pastebin?
14:24.40_ShrikEriddlebox: I missed it, could you paste the link again?
14:26.18riddleboxsure, let me get it
14:26.48riddlebox_ShrikE, http://pastebin.ca/924527
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14:39.13_ShrikELooks ok, my experience with SLA is pretty limited though
14:40.59riddlebox_ShrikE, the SLAStations are your actual extensions right?
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15:15.56dacstrying to config astercrm , the eventsdaemon is asking for my * username and secert...where can i get those , i really forgot them
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15:20.35kamitodoanyone familiar with SPA2102?
15:25.28kamitodohow do I tell if it is locked?
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15:45.45jameswf-homepong
15:46.19_ShrikEkamitodo: have you tried logging in with the info in the manual?
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15:55.23kamitodo_ShrikE: I have no manual.  Bought a used one off amazon.
15:55.38_ShrikEyou can download the manual from linksys
15:56.50kamitodolooks like both the factory reset and user factory reset command are password protected.   does this mean it's locked to a specific ITSP?
15:57.30*** join/#asterisk hi365 (n=hi365@213.151.52.239)
15:57.36jameswf-homeyou can usualy crack em open and do a hard reset
15:58.05jameswf-homeif its a vonage box that has connected to the internet it is probably hosed
16:00.24kamitodonot sure if it's vonage.  it makes connections to the following sites:
16:00.27kamitodozagbot.com
16:00.28kamitodopatbox3.patrickdk.com
16:00.28kamitodomighty.proclabs.net
16:00.28kamitodo64.34.245.224
16:00.33kamitodoserver.donkeyfly.com
16:00.36kamitodowww.innomedia.com
16:00.56*** part/#asterisk jivco (n=jivco@85.187.217.6)
16:02.19kamitodojameswf-home: what do you mean by crack them open?
16:03.56jameswf-homelike pop out the screws
16:05.13MatBoyIs there a sipserce out there where I can connect on using my nic cards to test my trunk ?
16:05.46MatBoyor is this a weird question ?
16:09.39*** join/#asterisk RoyK (n=roy@ti211110a081-7023.bb.online.no)
16:09.54*** join/#asterisk RoyK (n=roy@ti211110a081-7023.bb.online.no)
16:13.19kamitodojameswf-home: i see.  i'm googling for a how-to on this, but finding nothing...
16:13.22*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
16:13.28*** join/#asterisk lisandropm (n=lisandro@190.1.22.76)
16:18.29*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
16:20.02dacsis there a channel for astercrm
16:20.12dacs!astercrm
16:20.25MatBoydacs, nothing to find on google ?
16:20.40jameswf-home~astercrm
16:20.57*** part/#asterisk lisandropm (n=lisandro@190.1.22.76)
16:20.58dacsMatBoy: i found the webpage but i have a question in the installation part
16:21.48robmac67kamitodo: plug in a handset dial **** to access the ivr & then 73738 to do a factory reset
16:22.00MatBoydacs, I'm very new her also, I have visited this channel more, but I'm into VoIP now totally... setting up a2billing
16:22.27outtolunchttp://forums.astercrm.org/
16:23.34dacs!ivr
16:23.38dacs~ivr
16:23.39jbot[ivr] Interactive Voice Response
16:23.46MatBoyhow do Atcom cards perform, recognized as Digium ?
16:24.02MatBoydacs, hehe, I wanted to know that one too :)
16:24.03kamitodojameswf-home: i opened the damn thing but there's no obvious reset.
16:24.22*** join/#asterisk qdk (n=qdk@195.242.194.42)
16:24.34MatBoykaldemar, can' t you look for the chipset how to manually reset it ?
16:24.40MatBoysometimes that is possible...
16:24.45MatBoyhardwired hack
16:25.16dacscan someone please guide me on where i can install and setup ivr with my *
16:25.37dacskaldemar: what are you trying to reset
16:26.39kamitodokaldemar?  guess you mean kamitodo?  I'm trying to reset SPA2102.  I don't have the admin pass for it
16:27.01robmac67<PROTECTED>
16:27.20kamitododid that. there's a password.
16:27.49jameswf-homejbot: tell dacs about buybook
16:27.50dacskamitodo: can you access the web application
16:28.29kamitodoyes.
16:29.26MatBoyno-one experience with atcom ?
16:29.31dacskamitodo: try user:user
16:29.43dacskamitodo: password:8995523
16:31.18*** join/#asterisk seanbright (i=seanbrig@65.207.74.18)
16:31.52dacskamitodo: is it virgin or SR SPA21202
16:32.24kamitodo8995523 is invalid
16:32.45dacs^^
16:33.02kamitodoalso invalid passwords are the ones posted here: http://www.dslreports.com/forum/remark,14450684~days=9999~start=1900
16:33.30kamitododacs: how do i tell if its virgin or SR?
16:34.48dacskamitodo: is it SPA2102-ACN or SPA2102-R
16:35.08kamitodoall I know is, upon startup, it's making connections (I'm monitoring via my router) to the following sites: zagbot.com, patbox3.patrickdk.com, mighty.proclabs.net, 64.34.245.224,server.donkeyfly.com, www.innomedia.com
16:35.59kamitodothe box says spa2102-na, the websites says: SPA-2102 Serial Number:FM500G308331 Software Version:5.1.12 Hardware Version:1.2.5(a)
16:36.27kamitodoClient Certificate:Installed
16:36.36dacskamitodo: get your self a good TFTP software and a proxy server and continue reading that forum, the answer to the quiestion is within
16:40.34drmessanoSPA-2102 with a password?
16:40.54dacsdrmessano: how are you sir
16:41.06drmessanoim ok
16:41.35kamitodoI'm afraid I'm lost.
16:42.06kamitodoshould I flush the firmware?
16:42.11drmessanoIs it from an ITSP?
16:42.23kamitododon't know.  bought it used.
16:42.29drmessanooh
16:44.19kamitodoif I buy a new one from amazon, are they unlocked?
16:44.27drmessanoYes
16:46.10[TK]D-Fenderkamitodo, Buy one from a normal reputable dealer and stop asking for trouble
16:46.39drmessano*NEW* is your friend
16:47.00dacs[TK]D-Fender: how are you doing sir
16:47.40drmessanoThe only NEW Linksys boxes that are questionable are PAP2s, anything else *NEW* from Linksys is going to be (I hate this word) unlocked.
16:48.09dacsdrmessano: unlock, unlock ,unlock
16:48.29dacs^^ Don't say UNLOCK ...errrr :)
16:48.45kamitodoalright.  thanks!
16:49.42ManxPowerhaving a password and being locked are NOT the same
16:50.19kamitodobtw, since i'm buying new, is the spa2102 the best thing of it's class?
16:50.43kamitodoManxPower: ok, what's the diff?
16:53.19drmessanoThe difference is whether or not the password or other restrictions are factory defaults
16:53.53drmessanoIf some programmed the box via tftp to disallow fact reset and other things, it's not technically "locked"
16:54.48drmessanoI'm not sure it really matters because locked, unlocked, et al are just a bunch of dumbass terms people invented to describe the boxes, and sadly have become common use
16:55.19kamitodoso if that't the case by flashing again via tftp i can remove the pass
16:55.55drmessanoIf the box is set to look at a specific tftp server, then no
16:56.13kamitodoaha.
16:56.52*** join/#asterisk jmacz (n=jmacz@190.90.35.212)
16:57.09drmessanoI suppose you can use a packet sniffer to see where it's looking for it provisioning file
16:57.17drmessanoThen fake a DNS server and a tftp server
16:57.34drmessanoMuch like the (oh god) unlocking trick for the PAP2
16:58.08kamitodooh boy.  i hoped not to have to do this
16:58.08ManxPowerkamitodo: A locked box can't have it's password reset by a factory default reset
16:58.26*** join/#asterisk RoyK_ (n=roy@ti211110a081-1633.bb.online.no)
16:59.21drmessanokamitodo: If it you can't fact reset it to even see if it's just provisioned, you're gonna need to do something
16:59.30ManxPowerTo me "locked" means, doing a factory reset doesn't clear out all the passwords.  Linksys/SIPura don't WANT to lock their boxes, but if they want to sell to the likes of Vonage or other companies that sell the hardware at a loss they must lock them
17:01.30kamitodoand what does provisioned mean here?  bound to a ITSP?
17:01.35drmessanoNo
17:01.43ManxPowerI know the universe will explode for saying this but this is one of the best explanations of "locked" I've seen in a long time: "drmessano: The difference is whether or not the password or other restrictions are factory defaults"
17:01.43drmessanoNot per se
17:02.56drmessanoProvisioned is just the act of configuring settings, in this case, via tftp
17:03.20*** part/#asterisk Paladine (n=paladine@ns2.scs-live.com)
17:03.26drmessanoWhen you start talking about binding, locking, and other bleh terms, those are usually hard coded in
17:03.59ManxPowerThere is some special key sequence like ##RESET that clears all setting back to their defaults on an non-locked box.  The SIPura docs will have the specific key sequence.
17:04.31drmessanoBut if, drmessano, feed a box a file to provision it, and set the field to block fact reset, TO YOU, it's as good as locked until you can reprovision it somehow
17:05.40drmessanoIt's called the GPP_K I believe
17:06.46*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
17:06.47kamitodook.  how does the packet sniffer help?  does the box try to contact their tftp server on startup?
17:06.57drmessanoIt could
17:06.59drmessanoListen..
17:07.26drmessanoWith the PAP2s, they were somewhat easy to (ARRRGH) unlock because there were LOTS of known constants
17:07.30ManxPowerkamitodo: for the amount of time you will spend on trying to hack a locked box you could buy several new ones
17:07.35*** join/#asterisk joelsolanki (i=joelsola@220.224.22.199)
17:07.47drmessanoTFTP addresses, FTP addresses, known GPP_K keys to reset, etc, etc, etc
17:07.58drmessanoYou have NO Idea of ANYTHING with this specific box
17:08.10drmessanoPut it back on eBay as "Some used box I had" and move on to a new one
17:08.40drmessanoEven if you DID manage to feed it a xml file..
17:08.54drmessanoWhich the chances of it not needing to be encrypted are slim as hell
17:09.12joelsolankiHi All
17:09.31drmessanoYou may open it up just enough to release it's got some good fact defaults from an ITSP and was purchased by them in that way
17:09.37drmessanorealize
17:09.41joelsolankiI am installing zaptel right now. zaptel-1.4.9.2 on centos 5
17:09.48kamitodoit won't be ethical to sell it that way.  i got it for $20 so no big deal.  (no wonder it was cheap!)  will just buy a new one.  let me know if someone wants the old one, I can just send it to you.
17:09.53joelsolankiI have installed kernel-devel in centos 5 already
17:10.11*** join/#asterisk angryuser (i=nononon@df01t2-213-44-91-185.d4.club-internet.fr)
17:10.21joelsolankii m giving ' make linux26 '
17:10.27joelsolankigrep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory
17:10.29joelsolankimake: *** No rule to make target `linux26'.  Stop.
17:10.35joelsolankiwhat could be the problem ?
17:10.45drmessanoUsed is used.. Throw it on eBay for $15 and put "ZOMG LQQK THIS IS LOCKED SPA-2102 ** UNLOCKERS SPECIAL **" and let someone else mess with it
17:10.47drmessanoReally
17:11.32joelsolankisymbolic link is already set
17:11.33joelsolankilrwxrwxrwx 1 root root 42 Feb 22 13:17 /lib/modules/2.6.18-8.el5/build -> ../../../usr/src/kernels/2.6.18-8.el5-i686
17:11.49joelsolankistill i m not able to install it on centos 5.
17:11.53joelsolankiany hints pl
17:11.54joelsolankipzl
17:11.56ManxPowerjoelsolanki: It would not hurt to try installing kernel-source
17:12.15ManxPowerjoelsolanki: don't do a make linux26 that is old and not needed
17:12.36ManxPowerunless you are running zaptel 1.0.x or something like that
17:12.47kamitodook, getting a new one.  thanks drmessano and everyone else for the advices.
17:12.50drmessanoyum install kernel-devel on CentOS 5
17:13.09joelsolankiyes i already install kernel-devel with ym
17:13.10joelsolankiyum
17:14.09drmessanoHave you updated kernel using something other than yum?
17:14.40ManxPowerjoelsolanki: what happens when you just do "make"
17:14.42joelsolankijust did ' make ' but it is telling that ' you do not appear to have the sources for the 2.6.18-8.el5 kernel installed
17:15.01drmessanoHave you updated your kernel using something other than yum?
17:15.07joelsolankilet me again try doing yum install kernel-devel
17:15.15*** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com)
17:15.21drmessanoHave you updated your kernel using something other than yum? ZOMG SHARKS
17:15.36joelsolankino i have not.
17:15.39drmessanoOk
17:15.40joelsolankii have used yum only
17:15.51drmessanoJust wanted to make sure you didnt create a version mismatch
17:16.12joelsolankihmm
17:16.22joelsolankijust did yum install kernel-devel
17:16.24joelsolankipackage kernel-devel-2.6.18-53.1.13.el5 (which is newer than kernel-devel-2.6.18-53.1.6.el5) is already installed
17:16.48joelsolankiis this a problem ?
17:17.28drmessanoDo a yum list kernel*
17:17.33joelsolankiok
17:18.01drmessanoMake sure the 2.6.XX-XX are the same all the way down
17:18.07joelsolankikernel.i686                              2.6.18-8.el5           installed
17:18.11joelsolankikernel.i686                              2.6.18-53.1.13.el5     updates
17:18.53drmessanosounds like you need to update the kernel to match the sources
17:19.03joelsolankioh.
17:19.17joelsolankiyum update kernel* ??
17:19.32drmessanoyeah
17:19.38joelsolankilet me do that
17:19.49drmessanoI'm not 100% sure about that.. but I have had this problem with that solution
17:20.03joelsolankigot it
17:21.41joelsolankii think i have to reboot the server one time  ?
17:21.49joelsolankikernel is installed therefore
17:21.50drmessanoI'm headed out for a bit.. I wanted to put my $.02 in about the kernel versions because i've gotten them out of sync before and had issues like yours.. i'm sure if that's not the ONLY problem, someone else a heck of lot better with Zaptel can help ya
17:21.56drmessanoYeah you do
17:22.00joelsolankiok
17:22.15drmessanottfn
17:22.23joelsolankiok
17:25.28*** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote)
17:25.33*** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl)
17:28.06joelsolankiok updating kernel worked !!
17:28.08joelsolanki:)
17:29.48joelsolankigoogling on ss7 found that chan_ss7 is the best to use open source though they have stopped the development.
17:30.23joelsolankiI have sangoma A104D. working on integrating sangoma A104D + ss7 + asterisk
17:35.00*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
17:38.23*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
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17:53.15*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
17:56.22TJNIIWhat do the sip Parse_srv messages mean?  I'm not getting very far with google.
17:57.09TJNIIDNS?
17:57.54*** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com)
17:59.06ManxPowerSRV is a DNS method to automagically find the correct IP address for services (like SIP) for a domain
17:59.19ManxPowersrvlookup=no in sip.conf
17:59.24*** join/#asterisk lowlevel (n=Stuart@76.10.182.6)
18:00.25TJNIIAah, ok.  That's what I gleaned from the source.
18:00.40TJNIIIt's not causing problems, I just didn't know what the messages meant.
18:02.36ManxPowerit might show down connections if you don't need it.
18:13.01lnxho there, do you know solotion for ringing detection in AGI with perl?  I want  to create  ,if ringing hangup, like script
18:14.09ManxPowerlnx: I don't believe there is a way.
18:14.47*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
18:15.10JunK-Ylnx: ur AGI must listen on the AMI port and listen to events.
18:15.27jmaczHi everyone, I'm having dropped incoming calls (27 hangup_cause on PRI debug). We've checked the line twice with the Telco guys and it's physically and data-link OK. If we connect the old PBX, none of the incoming calls are dropped. Any ideas what may be causing this behavior (we are using a te210P)???
18:15.32ManxPowerof course there are many kinds of "ringing"
18:16.26ManxPowerCause 27 is Destination Out of Order
18:17.00lnxManxPower JunK-Y okay ty. Where can i fond more info about it? the-asterisk-book.com?
18:17.05ManxPowerMaybe your have the wrong switchtype=  You could expect national, 5ess, and DMS to mostly work for any swotch
18:17.07jmaczManxPower, that's it, and it may be caused by physical data-link problems, right?
18:17.15ManxPowerlnx: We don't even know what type of ringing you are talking about
18:17.24ManxPowerjmacz: I've never seen that on INCOMING calls.
18:17.36jmaczManxPower, it's EuroISDN, and it's the only one I guess that fits our Telco
18:17.58lnxManxPower: well i have defined a callfile with  channel asterisk makes a simple call ...
18:18.18*** join/#asterisk steliosk (n=Stelios@79.131.73.109)
18:18.57jmaczManxPower, we've seen with the Telco guys that we send a PROGRESS and the receive an STATUS before the Telco sends us the DISCONNECT with the 27 hangup cause
18:18.59lnxManxPower: if the called endpoint is ringing ...
18:19.17ManxPowerlnx: Have the .call file send the call to Local/ then in the dialplan use a Dial with a very short timeout, then check the value of HANGUPCAUSE or DIALSTATUS
18:20.06lnxManxPower: thank you
18:20.10*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
18:20.29ManxPowerlnx: none of this will work on FXO (analog or T-1) signaled port.
18:20.44lnxManxPower: how can i get DIALSTATUS variable?
18:20.59ManxPower${DIALSTATUS}
18:21.10ManxPowerNONE of this requires AGI
18:21.27ManxPowerexcept for maybe the creation of the .call file
18:21.27lnxi see , thx
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18:22.42jmaczManxPower, may it be a chan_zap issue?
18:23.44ManxPowerjmacz: I have no idea.
18:24.17ManxPowerI would do a pri debug or pri intense debug, report it to bugs.digium.com.  Make sure there is information from ONLY one failed call.
18:24.46jmaczManxPower, ok, I'll try that
18:24.54jmaczthank you very much
18:34.29*** part/#asterisk kamitodo (n=user@70.17.85.123)
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18:35.54nvrpunkI have a portech gsm 372, its working fine outbound now I am trying to get inbound working
18:35.58nvrpunkI get     -- SIP/1011-081b6030 is making progress passing it to SIP/sip10-0826c010
18:36.08nvrpunkwhen trying to call to it
18:36.30nvrpunki want it to route to 1012 which it is configured to do
18:36.34nvrpunkbut all I get is ringing
18:36.44nvrpunkand the device doesnt actually monitor its call progress
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18:38.34MatBoyis it possible to call internal without a trunk ?
18:39.05MatBoyso only on the LAN withotu having a trunk for this to the outside world ?
18:40.07ManxPower~trunk
18:40.08jbotsomebody said trunk was is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term.
18:40.18ManxPowerWe don't use the word trunk here.  Use the right term?
18:40.51*** part/#asterisk bkw_ (n=brian@adsl-70-234-169-126.dsl.tul2ok.sbcglobal.net)
18:41.01MatBoyManxPower, I used that term because I'm using a2billing and I have the idea that I need a " trunk" to be able to make a call anyhow
18:41.54TJNIIYou'd probably get a better answer if you describe in detail what you are trying to do.
18:42.09MatBoyok, if that's allright I will do that
18:42.12ManxPowerwe don't really support billing software here.
18:42.15MatBoyI don
18:42.25MatBoyI don't want to be annoying :)
18:43.45MatBoyI have ordered a Digium E1 card that is not here yet, so what I try to do is calling on LAN to test it already when I setup 2 customers and call as " friend" in the system using a prefix for internal calls
18:44.10MatBoymy question is if this will be possible so I'm not looking or trying to do something that can;t be done
18:44.52ManxPowerMatBoy: SIP?
18:44.54drmessano~book
18:44.54jbotAsterisk: 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, or see ~buybook
18:45.07MatBoyManxPower, yes
18:45.43ManxPowerMatBoy: you would edit sip.conf to set up the SIP peers and devices, then edit extensions.conf to set up the call routing.  What SIP phone are you using?
18:45.46MatBoyyep, I'm looking at every known document of course, I'm not a "ask for a solution"  person :)
18:46.14MatBoyManxPower, x-lite to test at the moment on 2 seperate computers
18:46.48*** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com)
18:47.43MatBoyManxPower, but the actuall question is maybe, will a2billing not manage this by default or do I always have to configure asterisk also for the a2billing part ? I mean, fir a2billing you need to include config files into the asterisk config files
18:48.11ManxPowerMatBoy: we don't know ANYTHING about a2billing here.  We deal with Asterisk, not 3rd party billing systems
18:48.22ManxPowerI suggest you use the correct a2billing support options
18:49.02MatBoyManxPower, I understand, but I was pointed to a2billing here some time ago, so I assumed people were using it also ? Just some global knowledge sharing ;)
18:49.24MatBoyManxPower, I'm already crawling their forum and docs a lot
18:50.02ManxPowerMatBoy: Best of luck
18:50.57MatBoyManxPower, yep, thanks ! but I don't want to act like I'm looking for a solution, only the right way where I might be question myself what direction I should take.. need some confirment in the beginning sometimes :)
18:51.04MatBoyI hope you know what I mean
18:51.44ManxPowerI think what you mean is that you are finding the a2billing support to be lacking so you are here in a desperate attempt to get help.
18:52.06MatBoyManxPower, no, not exactly
18:52.14drmessanoMatBoy: It sounds to me like you seriously need to work on your understanding of Asterisk before even attempting something like A2Billing which is going to require a LOT of assumptions due to its lack of documentation
18:53.34MatBoyManxPower, I'm trying to tell that if it's not allowed here to talk about 3rd party, I don't have a problem with it but just a question. Why not share info that is not 100% related when someone has it.. I almost do the same on other channels like vmware.. if someone has a HW question... why not ?
18:54.06MatBoydrmessano, yes there is missing some stuff indeed, but still, you need to start somewhere...
18:54.12drmessanoYes you do
18:54.13drmessanoThe book
18:54.15drmessano~book
18:54.15jbotAsterisk: 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, or see ~buybook
18:54.34MatBoydrmessano, yep, found that one yesterday, very nice 600 pages :)
18:54.37MatBoygood resource
18:54.40drmessanoThis channel isn't for "I just installed Asterisk, help me please callcenter".. This is more for intermediate issues
18:54.44drmessanoOk good, go read
18:55.15MatBoydrmessano, that is what I wanted to say... I'm not that kind of guy... I will be here all the time when I'm online... I can share my knowledge later on also...
18:55.19drmessanoAfter you have read, maybe some of the poorly documented pieces of A2billing will make sense
18:55.37drmessanoMatBoy: You are sounding exactly the opposite of what you are saying
18:55.46MatBoydrmessano, why would I be here all day already than ;) I should have quite already when my question was not answered :)
18:55.50drmessano"I am not the kind to not read and beg for help, but can you help me please"
18:56.02MatBoydrmessano, that's because I don't want to be rude, I hate those people also
18:56.14drmessanoGreat.. So don't be one..
18:56.17MatBoydrmessano, did I beg somewhere ?
18:56.20ManxPowerMatBoy: you are trying to amputate a diseased limb without knowing basic anatomy
18:56.26jameswf-home~rude
18:56.26jbotrude is, like, making me tell people things that they dont want to be told
18:56.26TJNIIMatBoy: Have you played with just a basic * server yet?  No billing stuff, just a virgin install?
18:56.57drmessanoYou were arguing with ManxPower over asking a question about an unsupported app and you clearly haven't got past Asterisk 101 yet
18:57.20jameswf-homeheh I can amputate a limb without basic anatomy as long as there are no expectations after the deed is done
18:57.20drmessanoYou're only making this harder on yourself
18:57.24MatBoyTJNII, before a little bit, but I'm actually discovering how such software would be a good additional thing to *, because when it isn't... you will think you can do stuff with it what never could be done.
18:57.25drmessanoLOL
18:57.45drmessano~nowwhat
18:57.46jbotSo you just installed asterisk and arent sure what to do now? visit http://www.a1b2c3.com/suilodge/metfun1.htm
18:57.56drmessano~now what
18:57.57jboti guess now what is 2*4?
18:57.57drmessanocrap.. where is it
18:58.09jameswf-homejbot spank MatBoy
18:58.09jbotACTION bends MatBoy over his knee and tatoos 'ibot' on MatBoy's pasty white buttocks.
18:58.14MatBoydrmessano, I think you like the bot ?
18:58.33*** part/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
18:58.40MatBoydrmessano, the simple question is... did you already know everything ? I guess not and I'm not saying I know anything at all
18:58.59drmessanoYou're asking about an app and you haven't gotten past Asterisk 101 yet
18:59.05drmessanoGO READ and stop ARGUING YOUR POINT
18:59.05jameswf-homeBetween me, jbot, and google I know everything
18:59.30drmessanoYou *ARE* being one of those annoying guys by continuing this line of insane arguing
18:59.37jameswf-home~troll
18:59.37jbothmm... troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or ...
18:59.40*** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com)
18:59.43drmessanoHave you even gotten two extensions calling each other yet?
18:59.49MatBoydrmessano, yes I will, but as you also have seen, a lot of people think it's simple to setup, and I have seen it's not... so I'm discovering what will be best
18:59.59nvrpunkwhat is the best way to encode wave files to g729?
19:00.08jameswf-homesimple is a relitive term
19:00.19drmessanoMatBoy: It will be simple after you.. ummm
19:00.28drmessanoRead?
19:00.29jameswf-homertfm
19:00.39MatBoydrmessano, ofcoruse, did I say it wasn't ?
19:00.46MatBoyno ;)
19:00.52drmessanoDude, stop trolling..
19:01.17MatBoyI'm not your dude, please stop trolling also... I think we have solved this than
19:01.26drmessanoGo back to your bridge
19:01.40drmessanoWith a copy of the book
19:01.55jameswf-home~dude
19:01.56jbotBe most excellent to each other!
19:02.06drmessanoTried to help and now, alas, I must terminate this communication
19:02.29MatBoydrmessano, don't tell me what to do, just give advice or don't say anything... that's looking like a wiseguy actually... sorry, but you are right, but the way you say is very.... uhm ;)
19:02.35TJNIIMatBoy: You don't actually have a problem.  You just don't understand how asterisk works.  Read the book and play with a clean * install.  You're not ready for a2billing.
19:02.40jameswf-homeMatBoy: did you try rebooting
19:02.59MatBoyTJNII, yes I will continue :)
19:04.43MatBoyTJNII, the book is very good... but the a2bliing part let it look likes it's automated.. like hosters that think they can admin servers when they can install PLesk.... but I wanted to know for sure if that's not it, because I know for sure it isn't :)
19:04.50drmessano:)))))) (((((((((:   ;)
19:06.30jameswf-homesimheh
19:06.37jameswf-home~fixit
19:06.37jbotto fix your issue follow these five steps... 1. Find a radio with a long cord, use an extension cord if needed to get a plug without a GFI. 2. Plug in the radio to the non GFI outlet. 3. Fill your bathtub with water. 4 bring radio with you and step into the tub 5. drop said radio. PROBLEMS SOLVED
19:07.40MatBoyjameswf-home, I think you never looked at discovery channel... that one was busted ;)
19:07.46drmessanoOk, back out to finish off my Magic Mystery Tour saturday.. I think the wife and I will stop at the A2billing store at the mall and get a book on pretzels
19:08.14drmessanoChowder
19:09.01jameswf-homeLies all Lies get a video camera and show me
19:09.45MatBoyjameswf-home, I think it;s somewhere on youtube if you want... water does not do such things very easy with power, it's much more a resistor
19:10.13jameswf-homeI saw the episode I believe it was a pool
19:10.51MatBoyjameswf-home, what I remember it was a bath with a multimeter and some stuff where they let a radio fall into the water from a shelf
19:11.15MatBoyvery simple explained
19:11.27MatBoyI never saw one with a pool, I don't stay home for it
19:11.42scooby2fsck a duck. Well its not my digium te212p as this sangoma a102 errors the same way. All zap calls stop. Errors like this: chan_zap.c: Ring requested on channel 0/2 already in use or previously requested on span 2.  Attempting to renegotiating channel.
19:12.07*** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote)
19:12.36*** join/#asterisk seanbright (i=seanbrig@65.207.74.18)
19:13.19ZPerteeI have an avaya pbx that I want to use with *.  What I want to do is connect an extension port on the avaya to my digium fxo card.  however that isn't working.  what signalling should I use?
19:14.43jameswf-homeok Episode 22 Bathtub Electrocution: CONFIRMED - virtually everything they dropped in the tub registered as a fatal shock
19:15.50jameswf-homewho stays up to watch tv I tivo and catch up saturday
19:18.08*** join/#asterisk Greek-Boy (n=email@41.221.58.4)
19:18.40MatBoyjameswf-home, ah ok, I thought the shock was to low, there was something they discovered that they didn't expect, that's what I remember
19:19.11MatBoyjameswf-home, what I know is that with a very simple and low shock you can mess up the rythm of you hard indeed
19:19.12jameswf-homeI think you can drop a radio into distilled water and live...
19:19.31MatBoyjameswf-home, let's ask them to test that comparing to this test
19:19.38MatBoywould be nice to see
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19:23.03pkwonghi all.
19:23.22jameswf-homeholy speak of the devil batman
19:23.29pkwonghehe.
19:23.33pkwonghey james
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19:39.01outtolunc..
19:41.02nvrpunkfor an incoming number of say 07906824351@1012
19:41.15nvrpunk_1012 should catch it correct?
19:41.53outtolunc1012 is a [1012] not an exten
19:42.26outtolunc@ domain/context
19:42.34nvrpunkok
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19:46.56tzafrir_homejameswf-home, ping
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20:00.13nvrpunki cant get this portech gdm372 to do incoming calls -.-
20:00.17nvrpunkno matter how hard i try
20:00.20dssman<PROTECTED>
20:00.39nvrpunkmeans its not finding the asterisk box
20:00.43nvrpunkis what it sounds like
20:00.51outtolunclook on the floor, is the cable laying there <G>
20:02.32dssmanlol
20:02.54nvrpunkso I add exten => _1012,1, Dial(SIP/sip1) for the portech gsm372 and it just rings and rings
20:02.58dssmanits server side I see the error... I can dial extensions, just the phone isnt registering one itself
20:03.14nvrpunkeven though the portech is showing 0790blah@1012
20:03.22nvrpunkas incoming to it
20:06.28nvrpunkcheck your configs
20:07.15dssmanif Im using asteriskNow, and I add a user, will that add extensions for me too?
20:07.31*** join/#asterisk rbd (n=rbd@216.148.216.55)
20:07.38linuxstbHi all.  I've set up a basic asterisk 1.4.18 system (hosted on a server located in a datacentre), and have various different kinds of SIP phones (Grandstream GXP-2000, a SPA3102, Ekiga softphone and Nokia E60) attached to my home network (behind a NAT firewall) for testing purposes.
20:07.41linuxstbThey can all dial each other, apart from the Nokia, which can make outgoing calls to the other phones, but refuses to accept any incoming calls - it never rings and seems to crash the Nokia SIP stack (I need to reboot the phone afterwards).
20:08.53rbdhi guys, I had a question about the stat function. the wiki docs on it are unclear. if I call stat(e, ...) for instance, what is returned if the file exists, and if it doesnt?
20:11.07jmaczdssman, check with *CLI> sip show peer <your_phone> if it's defined somewhere (like sip.conf)
20:13.53dssmanyea appears to b
20:14.46dssmanI hate waiting for the phone to reboot :P
20:15.38dssmanokay, gettin closer... now I have username / auth mismatch
20:15.41*** join/#asterisk SteveTotaro (n=root@pool-71-179-207-15.bltmmd.east.verizon.net)
20:15.56dssmanwhat is digest?
20:16.43dssmanand for every phone that is going to be connected do I need ot make an entry in the sip and exten .conf files?
20:17.11dssmanOMG I have a registered phone
20:18.25dssmanuch, now none of the demo extensions are working
20:18.27dssmangrr
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20:23.35dssmanohh, lol nothin works :D
20:28.15*** part/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
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20:39.16TJNIIdssman: You have a registered phone?
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20:41.18teknoprepi love polycom phones
20:42.10drmessanoI love little chickens and shiny things
20:43.17teknoprepwent on ebay and purchased a bunch of cheap ip 501's
20:44.22drmessanoThe famous shoretel phones?
20:48.55*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:49.04*** join/#asterisk SparFux (n=raoul@e182029172.adsl.alicedsl.de)
20:49.18TJNIII can never find polycoms cheap.
20:49.53SparFuxI have problems compiling mISDN for AMD64 on debian.
20:50.55SparFuxmISDN-1_1_7_2 doesn't compile. It says something needs to get fixed from CFLAGS to EXTRA_CFLAGS. And the git version compiles, but when activating capi it crashes. The whole system freezes.
20:51.37TJNIIThat doesn't sound like a compile problem.
20:53.34SparFuxWith the git version, it is not.
20:53.46SparFuxI don't know, wether it has something to do with AMD64 arch.
20:54.04drmessanoAlways safer to go with 32-bit
20:55.41SparFuxdepends. The NXbit is a nice feature busting almost all buffer overflow issues.
20:56.13drmessanoFeatures are great when some parts don't work well under 64-bit
20:57.17SparFuxyet
20:57.33drmessanoI hate that word
20:57.44SparFuxneway, I prefer the safe thing with some thing not working over the unsafe thing.
20:58.18*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
20:59.08drmessanoI think it's dumb buying 64-bit hardware for Asterisk when the same money can get you a decent multicore 32-bit and some expectation that various drivers for hardware will all work
20:59.18drmessanoBut thats my personal opinion..
21:02.01SparFuxWell, I just switched from my 32 bit system over to this 64 bit system.
21:02.06SparFuxright today :-)
21:02.41SparFuxAs long as SMP is in place, I will refuse to buy multicore. SMP is a bad idea.
21:03.00SparFuxAnd even worse is 8-core cpus.
21:03.41drmessanoHeh.. and buying a 64-bit cause "One day it will all work" is better? :)
21:03.54SparFuxI think it will work in one day.
21:03.59SparFuxexatly in one day :-)
21:04.05lnxi have  no idea how can i make a procedure does: call a number via callfile; if the endpoint is ringing hang up; else do blahh.
21:04.07SparFuxI am not even sure it is a 64 bit issue.
21:04.22riddleboxdrmessano, have you ever dealt with sla?
21:04.29*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
21:04.35SparFuxplus, it is the git version, misdn 1.2. It is not supposed to be stable.
21:04.37drmessanoSLA is evil, no
21:04.52riddleboxwhy is it evil?
21:05.26drmessanoI just don't believe in SLA or the Easter Bunny
21:05.51*** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
21:06.15lnxring detection is  impossibble :/
21:09.05jameswf-homeimpossible for who
21:11.25SparFuxGet this: http://linuxgazette.net/107/pramode.html
21:12.16*** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl)
21:12.51lnxjameswf-home: which variable contains this : logger.c:     -- SIP/0009*001-0959fb30 is ringing  ?
21:14.13jameswf-homelikely verbose....
21:14.23jameswf-homecouldnt say without looking
21:14.30SparFuxI get this strange error when compiling: scripts/Makefile.build:46: *** CFLAGS was changed in "/usr/src/mISDN/mISDN-1_1_7_2/drivers/isdn/hardware/mISDN/Makefile". Fix it to use EXTRA_CFLAGS.  Stop.
21:14.53[TK]D-Fenderlnx, youare going to have to complete recode your own dial process..... goo luck with that...
21:15.52seanbright-homeSparFux: what do the mISDN maintainers say?
21:18.43lnx[TK]D-Fender: what do you think, why asterisk hasn't a ringing status variable? BTW i can't recode.
21:19.12seanbright-homeSparFux: and didn't you already say that the git version works?  why are you using the release version?
21:19.35SparFuxgit version crashes when activating capi.
21:19.46seanbright-homeSparFux: ahh
21:20.33*** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote)
21:20.39tzafrir_homeSparFux, is CLAGS used explicitly there?
21:20.53SparFuxYes. It is.
21:21.09lnxit is weird there is no solution for call testing what analyse ringing  only  :>
21:21.15tzafrir_homeAdn you fixed it to use CFLAGS?
21:21.59*** part/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote)
21:22.54lnxjameswf-home: i mean variable like DIALSTAUTS.
21:23.01lnx*STATUS
21:23.33[TK]D-Fenderlnx, you can't do anything in the middle of a DIAL.
21:24.02lnxsaaad :(
21:24.28seanbright-homelnx: i missed the beginning of the question, what are you trying to accomplish exactly?
21:24.29*** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
21:24.49adeelis there a way to disable showing ALL manager connects?
21:24.54adeeloutside of hacking the source?
21:25.41seanbright-homei doubt it
21:25.49lnxseanbright-home: main :)  i have  no idea how can i make a procedure does: call a number via callfile; if the endpoint is ringing hang up; else do blahh.
21:26.24lnxthere is an  entry in full.log   logger.c:     -- SIP/0009*001-0959fb30 is ringing
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21:26.50seanbright-homelnx: you would have to modify the Dial application
21:27.14seanbright-homelnx: once you are in Dial, you can't do anything until the call leaves the application
21:27.18lnxmay a variable like DIALSTATUS contains  *  is ringing
21:27.41seanbright-homelnx: no, unfortunately there is no "out of the box" way of doing that
21:27.50seanbright-homelnx: you'll have to modify the source or have someone do it for you
21:27.50lnxdamn
21:28.09seanbright-homeadeel: what version are you running?
21:28.11*** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr)
21:28.26[TK]D-Fenderlnx, Why do you want to hang up if its rining?
21:28.46adeel1.4.8
21:29.10anonymouz666[TK]D-Fender, just curious, do you know if it's possible to 'reinject' a PRIO do caller into queue, once the caller is already waiting?
21:29.24*** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep)
21:29.38[TK]D-Fenderanonymouz666, nothing I'm aware of
21:29.40lnx[TK]D-Fender: because i have to run test calls ...
21:29.41adeelseanbright-home, i set displaymanagerconnects = no in manager.conf but that doesn't seem to be doing anything
21:30.26[TK]D-Fenderlnx, Sorry, unless you're prepared to do some serious recoding your idea jsut isn't workable
21:30.29lnxif u have an idea how to test automatically thet calls are works fine
21:30.38lnxplease share  with me :)
21:30.50seanbright-homeadeel: try 'displayconnects = no'
21:31.11adeelseanbright-home, i'll try
21:31.18lnx*that
21:31.23seanbright-homeadeel: make sure to come back and thank me
21:31.27seanbright-home;-)
21:31.52drmessanoIf it doesn't work, will you give him a Twinkie?
21:32.08seanbright-homeand it will work
21:32.15seanbright-homei'll bet my last twinkie on it
21:32.17seanbright-home...
21:32.21drmessanoSays he who has no Twinkies
21:32.38lnxi'm using Twinkle :)
21:33.32adeelseanbright-home, well now it just keeps saying == Parsing '/etc/asterisk/manager.conf'; Found
21:33.38*** join/#asterisk _matt (i=matt@2001:770:168:1:20b:cdff:fe04:843a)
21:33.45seanbright-homeadeel: but the connects are gone, yeah?
21:33.50adeelseanbright-home, yes
21:33.53adeelseanbright-home, so i guess thanks
21:33.57seanbright-homeheh
21:34.01drmessanoOhhhh
21:34.11drmessanoThat's a HALF-TWINKIE solution
21:34.14adeelhaha
21:34.22seanbright-homethe "Parsing" foo isn't coming from the manager, its coming from the configuration system
21:34.39adeelanyway to disable that?
21:34.51seanbright-homeset verbose to 1
21:34.58seanbright-homeerr
21:35.02seanbright-hometurn off verbose
21:35.04seanbright-home:)
21:37.08seanbright-homelnx: with the test you are building, what is the alternative to ringing?
21:37.15adeelseanbright-home, ah, well i need the verbose level when i'm debugging
21:37.22seanbright-homeadeel: then you're screwed.
21:37.49adeelseanbright-home, yeah i know...i wonder if there's a way to move the == Parsing statements to a different logging level/context ....e.g. debug or something
21:37.51seanbright-homeadeel: or delete line 826 of main/config.c
21:38.18adeelline 826 huh? i'll take a look at it
21:38.18adeelthanks
21:38.20seanbright-homeadeel: change 826 from:
21:38.21seanbright-homeast_verbose(VERBOSE_PREFIX_2 "Parsing '%s': ", fn);
21:38.23seanbright-hometo:
21:38.38adeelnow if only i could get call pickup working right....
21:38.43seanbright-homeast_log(LOG_DEBUG, "Parsing '%s': ", fn);
21:38.51seanbright-homeand you're golden.
21:39.09adeelseanbright-home, oo, very interesting
21:39.29seanbright-homethen create a patch file so you can do it again when you upgrade
21:39.58adeelyeah, i have to create a patch file anyways...it's easier using my package manager (i'm on gentoo)
21:40.09seanbright-homegotcha
21:40.20seanbright-homewell good luck and god speed.
21:40.36*** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net)
21:40.53drmessanoROFL...
21:40.54adeelseanbright-home, any experience with call pickup?
21:41.04seanbright-homeadeel: no sir
21:41.12drmessanoIm watching the CLI as a buddy of mine is getting his wakeup call for work
21:41.22drmessanoHe keeps failing the math question from Allison
21:41.31seanbright-homethere's a math question?
21:41.33adeelwhat's the question?
21:41.36seanbright-homeis that to ensure you're actually awake?
21:41.50drmessanoYes
21:41.58drmessanoI have a somewhat modified version
21:42.13drmessano-- Playing 'wrong-try-again-smarty' (escape_digits=) (sample_offset 0)
21:42.16lnxseanbright-home: i'm out sorry
21:42.23lnxgo to sleep :P
21:42.37lnxthanks the help
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21:43.46seanbright-homelnx: if all you care about is "failure"
21:43.54seanbright-homeyou could just set a short timeout on the dial
21:43.59seanbright-homelike, say, 5 seconds
21:44.19drmessanoHmm
21:44.31seanbright-homethen check DIALSTATUS when you get back from Dial(), and if its CANCEL then it timed out
21:44.43seanbright-homeotherwise you could assume it rang
21:45.18seanbright-homeits also possible that wouldn't even come close to being correct
21:45.20seanbright-homebut its a thought.
21:46.14drmessanoROFL
21:46.18drmessanowow
21:46.32drmessanoI think he left his phone on Auto-Answer....
21:47.07drmessanoHe's been on the phone with Allison for 10hours and 20minutes
21:47.10nvrpunkcan rasterisk convert all files in a directory?
21:47.27nvrpunkwith one command
21:47.34hi365_many idea what this error means?
21:47.34hi365_m[Mar  1 23:46:16] NOTICE[3791] chan_sip.c: Unable to create/find SIP channel for this INVITE
21:47.34hi365_m[Mar  1 23:46:16] WARNING[3791] chan_sip.c: sip_xmit of 0xeab194 (len 486) to 212.150.88.20:5060 returned -2: Network is unreachable
21:47.55nvrpunkreinvite=no
21:47.56seanbright-homenvrpunk: no sir
21:48.16nvrpunkseanbright-home, anyone mind helping me with this script then?  for I in *.wav; do convert -OPTIONS "$I" "$I".g729; done
21:48.17seanbright-homenvrpunk: can't use sox?
21:48.21jqlfor i in dir/*; do rasterisk ...; done
21:48.24jqlvoila
21:48.25nvrpunkdoes sox do g729?
21:48.28jameswf-homemy sox are dirty
21:48.35*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:49.03jqlwell; do rasterisk ... $i; done
21:49.40drmessanoI scrwed up
21:49.46*** join/#asterisk amir_ (n=amir@unaffiliated/amir)
21:49.51drmessanoI did chan_spy, but I should have transferred the call
21:49.52seanbright-homenvrpunk: what is 'convert'?  never used it
21:50.05nvrpunkseanbright-home, convert = rasterisk it was like foo
21:50.11*** part/#asterisk amir_ (n=amir@unaffiliated/amir)
21:50.13seanbright-homeohhh
21:50.20nvrpunkjust mean convert using whatever tool
21:50.35*** join/#asterisk vrtk (n=bruno@201009059057.user.veloxzone.com.br)
21:51.41seanbright-homefor a in /path/to/files/*.wav; do asterisk -rx "file convert $a `echo $a | sed -e s/\.wav$/.g729/g`"; done
21:51.45seanbright-homethat might work
21:51.57seanbright-homeand i emphasize the word _might_
21:52.13hi365_mnvrpunk:was that for me (reinvite-no)?
21:52.31nvrpunkhi365, yes
21:52.57riddleboxweird, I upgraded to 1.4.18, and now it seems that the system is seeing the sla stuff
21:53.33nvrpunkfor i in *; do rasterisk -x "$i" "$i".g729"; done
21:53.33nvrpunk<PROTECTED>
21:53.42seanbright-home?
21:53.48seanbright-homewhat was unclear about:
21:53.49seanbright-homefor a in /path/to/files/*.wav; do asterisk -rx "file convert $a `echo $a | sed -e s/\.wav$/.g729/g`"; done
21:54.26jameswf-homeescape the spaces
21:54.27hi365_mnvrpunk: didnt work
21:54.41nvrpunkhi365, no clue then
21:54.43hi365_mcan having the hosname settings wrong have anything to do with that error?
21:56.41nvrpunkseanbright-home, that gives me unable to open input file
21:56.49nvrpunksec
21:57.08hi365_myup - wrong hostname!
21:57.10riddleboxawesome, now sla is working, I wonder if there was just a bug in 1.4.17
21:58.53nvrpunkseanbright-home, yeah, unable to open input file
21:58.55nvrpunkweird
21:59.00seanbright-homehmmm
22:00.00seanbright-homenvrpunk: this works for me
22:00.02seanbright-homefor a in /var/lib/asterisk/sounds/*.wav; do b=`echo $a | sed -e 's/ /\\ /g'`; rasterisk -x "file convert $b `echo $b | sed -e s/\.wav$/.g729/g`"; done
22:00.46nvrpunkhmm
22:00.55nvrpunkmaybe asterisk doesnt do input wav*
22:01.31seanbright-homei just said it works for me
22:01.46seanbright-homei just converted all of my wavs to gsm (don't have g729)
22:02.12jameswf-homehttp://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
22:02.23nvrpunkjameswf-home, already read that
22:02.32nvrpunki used germanix before
22:02.41nvrpunkbut get a slight glip at the end of every file
22:02.42seanbright-homenvrpunk: can you pastebin the output you are seeing?
22:02.46seanbright-home~pb
22:02.47jbot[~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
22:03.01nvrpunkUnable to open input file: /root/soundset/sounds/conf-hasleft.wav
22:03.05nvrpunkfor every file
22:03.09nvrpunkno need to pastebin
22:03.16seanbright-homeasterisk running as root?
22:03.22nvrpunkseanbright-home, yes
22:03.39nvrpunki even chmod 777 the files
22:03.58seanbright-homewtf?
22:04.05jameswf-homewhat was wrong with s'ox
22:04.14nvrpunkdoes sox do g729
22:04.15nvrpunki asked
22:04.33nvrpunkseanbright-home, was trying to see if quality is better using rasterisk and official digium codec
22:04.55seanbright-homenvrpunk: not based on a quick google, but i could be wrong.
22:05.03*** join/#asterisk philipp64 (n=chatzill@pool-71-112-32-245.sttlwa.dsl-w.verizon.net)
22:06.11seanbright-homenvrpunk: what version of asterisk?
22:06.35nvrpunkseanbright-home, its converting the gsm's I have
22:06.43nvrpunkmust be something with the wav sample rate
22:06.47nvrpunkthat asterisk isnt converting
22:07.04seanbright-homenvrpunk: gotcha.  so you're good or no?
22:07.12nvrpunkgood ;)
22:07.16seanbright-homegood.
22:07.16nvrpunkmany thanks
22:07.20nvrpunknice script btw
22:07.23seanbright-homeheh
22:07.40seanbright-homei write all my production code in bash
22:08.17nvrpunkheh
22:08.33nvrpunkmy boss asked me how to set the proxy to not load a gui
22:09.52jameswf-hometit tit tit ta tat atta tatt BAM there it is
22:10.07*** join/#asterisk Phillhun1 (n=PHunt@3.72.233.220.exetel.com.au)
22:12.49Phillhun1Hi I have a question I have an NEC IPK2 and a sip trunk card that I want to connect to a Cisco call manager but it looks like i need a Sip Proxy or Registrar server to go between the 2 can anyone help
22:13.06nvrpunkhmm
22:14.28*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
22:14.33nvrpunkseanbright-home, conversion sounds like crap :/
22:14.36nvrpunkcompared to the other
22:14.37nvrpunkhaha
22:14.40seanbright-homeheh
22:14.46seanbright-homewell then it was all worth it
22:15.04nvrpunkgermanix did better quality but theres a little noise at the end of each file
22:16.19ManxPowerPhillhun1: Perhaps you should ask on a channel that has to do with the software/hardware you are using.  This is not a general VoIP channel.
22:16.35*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
22:17.30ThatKidKelis there a way to disable music on hold for a particular call
22:17.34*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:17.34*** mode/#asterisk [+o russellb] by ChanServ
22:17.38Phillhun1Hi I was hoping that someone would no if Asterisk could be setup as a Sip proxy/Registrar server
22:17.41ManxPowerThatKidKel: yes.
22:17.55ThatKidKelManxPower..  Could you point me to some documentation or give me the command..
22:18.03ManxPowerPhillhun1: You would use the call manager running SIP as the sip proxy/registrar
22:18.09[TK]D-FenderThatKidKel, Set it to a class that has nothing to do.
22:18.30ManxPowerThatKidKel: you have to enable MoH using the "m" opton to Dial.  So no MoH is the default.
22:18.32ThatKidKelPhillhun1..  Check into Openser..
22:19.00ThatKidKelManx.  I have no options on my dial command; and its still giving me music..
22:19.14ManxPowerThatKidKel: then you don't live in the same universe as the rest of us.
22:19.32ManxPowerperhaps you don't mean MoH when dialing, but MoH when a caller is on hold?
22:19.48ManxPoweror maybe you mean MoH when a caller is in a queue?
22:20.07ManxPowerI can only guess at what you mean.
22:20.12Phillhun1apparently Call manager doesn't do sip proxy
22:20.28ManxPowerPhillhun1: neither does Asterisk, but you don't need a REAL sip proxy.
22:20.30ThatKidKelManxPower..  I have no options on my dial command..  And when a call is placed on hold, it starts palying music..
22:20.40ManxPoweryou just need a place to send calls to, Cisco will do that.
22:20.55ManxPowerThatKidKel: set the musiconholdclasss to something that does not exist
22:21.49ThatKidKelI'm letting my calls from the PSTN hit Asterisk instead of OpenSER, and I must say, I kinda like it.
22:22.04Phillhun1the problem is the NEC will only talk to a sip proxy
22:25.38ManxPowerPhillhun1: the NEC CANNOT know if it is talking to a SIP proxy or a SIP registrar or a SIP B2BUA
22:25.56ManxPowerAll it can know is that it sends SIP packets to a destination that doesn't reject them
22:28.53ManxPowerPhillhun1: pretty much all SIP Phones say they need a proxy
22:31.06Phillhun1I spoke to an NEC engineer who said they are only able to talk to a Sip Server / Proxy
22:31.45chavignyyo
22:33.05robmac67NEC in the UK use Grandstream ATA's
22:34.10*** join/#asterisk esaym (n=user@72.183.198.134)
22:44.05nvrpunkseanbright-home, sln to g729 == good gsm to g729 == bad
22:44.18seanbright-homenvrpunk: well that makes sense
22:44.26seanbright-homesln is uncompressed
22:44.32nvrpunkaye
22:44.33nvrpunk:)
22:44.41Qwellgsm to g729 == terrible
22:44.46nvrpunk<-- paid for custom prompts
22:44.54nvrpunkbut they didnt provide g729
22:44.56jqlthe other way sucks too
22:45.18nvrpunkso, does anyone have any experience with the portech gsm372?
22:45.30nvrpunkive spent a couple of hours trying to get inbound to work on it
22:45.37nvrpunkoutbound works great
22:45.46jqlwhat happens inbound?
22:45.56nvrpunkit just rings and rings
22:45.56nvrpunk:)
22:46.03jqlyou should answer it
22:46.03nvrpunkdoesnt seem to route anywhere
22:46.07nvrpunkhaha
22:46.13nvrpunkit doesnt ring any phone
22:46.23jqlnot even asterisk?
22:46.28jqlasterisk can answer phones
22:46.39nvrpunkhmm
22:47.33nvrpunk<PROTECTED>
22:47.34nvrpunk<PROTECTED>
22:47.39nvrpunkthats all i see in asterisk
22:47.48*** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:48.30jqlwell, you have a problem there
22:48.42nvrpunkon the gsm gateway I see the inbound as    0790XXXXXXX@1012
22:49.14nvrpunkjql, im passing out one sim out the gateway asiacell, and trying to call  back in
22:49.14jqlbut the gateway never feels the urge to answer?
22:49.32nvrpunkjql, the gateway sees the incoming call
22:49.36nvrpunkjust nothing happens with it
22:49.39jqlsees it indeed
22:50.02jameswf-homealliot of gsm gateways use revpol
22:50.11jameswf-homeallot
22:50.36nvrpunkhttp://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
22:50.41nvrpunkthats the gsm gateway i haave
22:50.49nvrpunki used it's context for the incoming
22:51.18nvrpunkive also changed it and tried making it just Dial(SIP/sip3,30)  a phone i have
22:52.07ManxPowernvrpunk: that is all you SHOULD expect until the far end answers.
22:52.19ManxPowersetting callprogress=yes would certinally screw things up
22:52.54ManxPowerbut I don't think that's the case there, since I don't believe chan_sip supports callprogress=yes  (this is fake call progress, not the reall call progress devices provide)
22:53.32nvrpunkManxPower, yes but chaging it to just Dial(SIP/sip3) i never get a ring on my phone
22:53.51nvrpunkwhich i would imagine the gateway should just link up and dial the phone
22:54.04ManxPowernvrpunk: there is no difference between 30 and no 30 except for asterisk timing out after 30 seconds
22:54.13nvrpunkyeah got that
22:54.18nvrpunkand going to step 2
22:54.20ManxPowernvrpunk: yes, it should.  contact the gateway's tech support
22:54.27ThatKidKelif Asterisk is out of the media path--is there anyway to put it back in?
22:54.35nvrpunkManxPower, all asian people who prolly wont help :)
22:54.55ManxPowernvrpunk: I doubt any of us can help either.  It sucks to be you.
22:55.13ManxPowerall people here can do is suggest random things that won't work, as the issue is IN THE GATEWAY.
22:55.44jameswf-home~random
22:56.52jameswf-home~random
22:57.03cesar_CRhello guys who knows the DMLink cards ???
22:59.18jameswf-home~clones
22:59.24jameswf-home~clone
22:59.25jbothmm... clone is a clone card - i.e. a worthless, unreliable piece of junk.  Is all that extra headache really worth the few dollars you're about to save?
22:59.54jqlbut... I get to save the money *now*
23:00.10cesar_CRok
23:00.31jameswf-homewell you save all sorts of money if your time is worthless
23:00.56jqlmy time is obviously worthless. *points at boss*
23:02.06jameswf-homeIf your an hourly employee encourage your boss to go cheap
23:02.14MatBoyI don't see any verbose messages in the cli when I add this to the logger for cli and start the cli with many v's
23:02.55jqlthe cli's -vs matter little
23:02.55*** join/#asterisk comprookie2000 (n=comprook@adsl-065-012-210-216.sip.bct.bellsouth.net)
23:03.02jqlcore set verbose 10 or whatever
23:03.07MatBoyok
23:03.11riddleboxman I am so excited now I can tell me boss that I can do sla and we will have a happy customer in a couple weeks when we install their system
23:03.27jameswf-homeyou need to properly setup logger.conf
23:03.45MatBoyjameswf-home, seems to be OK for what I find online
23:07.59ManxPowerriddlebox: have you actually TESTED SLA?
23:08.56riddleboxManxPower, I just tested it with 1 line, hopefully next week I can get into the office to test on 4 lines
23:13.30jameswf-homeUnable to register tone zone 'us' means something pooched right?
23:18.38*** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
23:20.02tzafrir_homeright
23:20.32tzafrir_hometry the latest zaptel (1.4.9.2) .
23:20.43tzafrir_homeOr just the memset patch to tonezone.c
23:20.51tzafrir_homejameswf-home, ==^
23:22.57jameswf-homeknown 1.4.8 bug? this is an install from those lime green guys..
23:25.12drmessanoMmm Lime Jello
23:25.48jameswf-homeknown 1.4.8 bug? this is an install from those lime green guys.. i am like flippin retarded
23:25.57jameswf-homebah
23:26.03jameswf-homeproves my point
23:26.22jameswf-homeis there a bug id i am like flippin retarded
23:29.38*** join/#asterisk friedrich| (i=friedric@trem-servers.com)
23:38.46*** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
23:39.04*** part/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
23:39.54*** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
23:40.05ManxPowerjameswf-home: if you were on the mailing lists you would know this already.
23:40.18docelmoAnyone in here have any experience with call files and the CDR's they write?
23:42.47*** join/#asterisk NiklasH_work (n=niklash@triton.dsv.su.se)
23:43.58jameswf-homeheh I get 400 emails a day from the list I look at keywords...
23:44.19NiklasH_workhi, i hope i'm in the right channel: I have a problem using a phone adapter for outgoing calls via asterisk behind NAT. Inbound calls work fine, but outbound give a busy tone right after the first connect. Anyone have any ideas as to what could be wrong?
23:44.43teknoprep~nat
23:44.44jboti guess nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
23:45.00teknoprep~sip_nat.conf
23:45.12NiklasH_worka software client works fine, so i don't think the nat conf is wrong
23:45.16teknoprepi don't remember which one it is
23:45.29teknoprepis it asterisk that is having problems ?
23:45.33teknoprepor is it your sip client
23:46.01NiklasH_workyes, just the adapter. softphone works fine.
23:46.14NiklasH_worki've tried two different adapters, none works
23:46.18teknoprepis the adapter on the same subnet as your asterisk box ?
23:46.31NiklasH_workyes, all on the same network
23:46.33dacs~astercrm
23:47.10jameswf-homeonly bug in my email is loadzone=au says works as loadzone=us
23:47.33NiklasH_workfrom the debug info from asterisk, it seems that the adapter places the call ok, then switches to busy after the ok from asterisk
23:48.35ManxPowerjameswf-home: what card do you have again?
23:48.47dacsis there is a channel for astercrm
23:49.49ManxPowerRe: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
23:50.29ManxPowerNiklasH_work: you must disable all NAT features on the adapter.  Asterisk's nat features handles all that stuff
23:50.55NiklasH_workiĆ've tried doing that, but i could have missed something. iĆ'll check again
23:51.20jameswf-homeno cards in at this point just loaded on vmware
23:51.31jameswf-homeztdummy
23:51.38*** join/#asterisk amir_ (n=amir@unaffiliated/amir)
23:51.39*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
23:51.39*** mode/#asterisk [+o codefreeze] by ChanServ
23:51.46ManxPowerAh, so it's not the bug I was thinking of.
23:51.52adeelhow do i get * 1.4 to record the cdr's in mysql?
23:51.56adeelis there an addons package?
23:51.59*** part/#asterisk amir_ (n=amir@unaffiliated/amir)
23:52.11ManxPoweradeel: yes
23:52.13jqlthere is
23:52.20jameswf-homeI am just curious if its 1.4.8 or the green people
23:53.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
23:53.46adeelManxPower, ah, thanks
23:54.15*** join/#asterisk JT (n=j@unaffiliated/jt)
23:54.28*** join/#asterisk brut- (n=brut@66.173.4.254)
23:57.34drmessanoWhere is host=blah&blah documented?
23:58.32drmessanoWasn't aware that & was a valid option, and need to know how multiple hosts are handled, in this case, with a peer to an ITSP
23:59.28adeeldrmessano, i always thought you had one host per host= entry, but can have multiple host= entries
23:59.43adeele.g., host=foo
23:59.45adeelhost=bar

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