IRC log for #asterisk on 20081116

00:00.29mchoulesouvage: please stop "helping" me
00:00.54mchoulesouvage: it's clear you have no idea what you're talking about
00:01.08mchoulesouvage: thanks but no thanks
00:02.28lesouvageIs there any second opnion, I really don't think I'm talking nonsense.
00:02.48mchoulesouvage: You are jabbering NONSENSE
00:03.12interfaithquestanyone tried chan_gtalk ?
00:04.13mchouinterfaithquest: I have it active but I've never tried it.  Wanna test?
00:04.23interfaithquestok
00:04.26lesouvagemchou: I could pastebin you a working example of what you are trying to achieve. But it seems that you are not interested.
00:05.38mchoulesouvage: no, I'm certainly not interested in your advice since you don't even know what I'm talking about
00:07.02*** part/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com)
00:09.05*** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au)
00:10.30lesouvagemchou: btw: why not simply test it so your own test would give the answer to your question.Just after the privacy() line add " exten => s,n,NoOp(the caling umber is: ${CallerID(num)}) (cahnge s if needed)
00:11.31*** join/#asterisk synchris (n=synchris@athedsl-156469.home.otenet.gr)
00:11.34lesouvagemchou: have a good live
00:11.54*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
00:11.54*** mode/#asterisk [+o russellb] by ChanServ
00:12.15lesouvage.
00:12.26raasdnil[TK]D-Fender: hey, phones are all woking good.  Got an issue with the fax machines.  Problem is the user dialed 1414001181339153100 on the fax and asterisk is trying to dial: 1343194105031100,1.  See http://www.pastebin.ca/1257290 for CLI output and extensions.conf
00:12.46*** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
00:13.13raasdnilsorry, http://www.pastebin.ca/1257291
00:14.06[TK]D-Fenderraasdnil: -- Executing [1343194105031100@out-nec:1] Dial("DAHDI/31-1", "DAHDI/g2/1343194105031100,,Tr") in new stack
00:14.13[TK]D-Fenderraasdnil: -- Called g2/1343194105031100
00:14.19[TK]D-Fenderraasdnil: there is no ",1" in there
00:14.39raasdnilyeah, but on the fax, I personally dialed 1414001181339153100
00:14.55raasdnilsomehow getting mangled.... :/
00:15.41raasdniloh... try the second pastie... i posted the wrong bit of extensions.conf (been up for a few hours now)
00:15.41[TK]D-Fenderraasdnil: That would be so mangled that I wouldn't trust your telling me that it happend at first glance
00:16.01raasdnil[TK]D-Fender: I didn't believe the person either.
00:16.02[TK]D-Fenderraasdnil: new PB please... make it the right one this time
00:16.19raasdnilhttp://www.pastebin.ca/1257291
00:16.40raasdnil[out-nec]
00:16.40raasdnilexten => _X.,1,Dial(DAHDI/g2/${EXTEN},,Tr)
00:16.52raasdnilon the end
00:17.09[TK]D-Fenderraasdnil: again I see no ",1" on the end of your dial
00:17.30raasdnilone sec
00:17.31raasdnilsorry
00:18.16raasdnilhttp://www.pastebin.ca/1257296
00:18.20raasdnilthat one is correct
00:20.33Dr-Linux|home[TK]D-Fender: tried alot but providing dial tone in AGI didn't work
00:20.57[TK]D-FenderDr-Linux|home: ok/fine/sure
00:21.11*** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
00:21.41[TK]D-Fenderraasdnil: Look in that PB.  still no ",1" as part of the dial.
00:21.58[TK]D-Fenderraasdnil: Put. down. The. Crack. Pipe. (c) JerJer
00:22.15raasdnil[TK]D-Fender: but it's the only thing keeping me awake! :)
00:22.31[TK]D-Fenderraasdnil: I would never do a drug named after a part of my ass <-
00:22.35raasdnilisn't the exten => _X.,1, the ,1 you are talking about :)
00:23.05raasdnilhas some basic misunderstood here
00:23.09[TK]D-Fenderraasdnil: "I'm" talking about?  I'm not talking about any of this.  YOU are the one says * is adding cahrs to a DIAL statement.
00:23.16[TK]D-Fenderchars*
00:23.43[TK]D-Fenderraasdnil: Paste the SINGLE line where you see * doing DIAL with those added chars
00:25.11raasdnil[TK]D-Fender: oh... <takes stupid cap off> I get what you are saying.  Lemmie go looks some more.
00:25.37[TK]D-Fender:)
00:25.54[TK]D-FenderNEXT!@@!@!!@ (c) BKW
00:26.26interfaithquestjoin #asterisk-dev
00:26.32interfaithquestoops
00:27.08raasdnilthrows [TK]D-Fender another newbie
00:27.26[TK]D-Fendergoes to hide the bodies
00:28.08raasdnilnewbie: [TK]D-Fender my asterisk just doesn't work.  It worked before, it doesn't work now.  I didn't change anything except 10 lines in the extensions.conf file. could that have something to do with it?
00:28.39[TK]D-Fenderreaches for his katana....
00:30.30raasdnil[TK]D-Fender: ok I think it might be the fax machine.
00:30.36raasdnilthe ,1 was a red herring
00:30.58raasdnilI dial 96927300 on the fax, the NEC prepends 1414 and into the * box comes:
00:31.10raasdnil<PROTECTED>
00:31.32[TK]D-Fenderraasdnil: Seems to truncate 2 digits
00:32.00raasdnilif you look at 1241439673 you can see the 1414 and the 96927373 interlaced... with the 969 chopped off the front and the 73 off the end
00:32.07raasdnilyeah...
00:32.08raasdnilweird
00:32.19raasdnildo you have to treat fax machines differently?
00:32.37[TK]D-Fenderraasdnil: this is your PBX being retarded... "not our problem"
00:32.39raasdnilmaybe I'll go put a POTS phone on that fax line and dial and see if the NEC system is playing funny buggers
00:32.43raasdnilheh
00:33.03raasdnilnewbie: my pabx company said that the asterisk guys would know how to integrate though... :D
00:33.26[TK]D-Fenderraasdnil: What else did your Rice Crispies say to you? :)
00:33.43raasdnilnyah... read it off the back of a pack of wheetbix :)
00:33.48raasdnilbbs
00:35.19harry_v90s in cali = perfect for fires.
00:35.50harry_vI wonder how fast * could dial call files such as advertising evacuation.
00:36.35*** part/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com)
00:37.32drmessanoThose services are mostly unreliable
00:39.15harry_vwhen it involves asterisk right?
00:40.58*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
00:43.31*** join/#asterisk shinao1 (n=shinao1@62.173.48.42)
00:49.00drmessanoNo
00:49.07*** part/#asterisk seaq (n=seaq@98.227.60.190.host.ifxnetworks.com)
00:52.40[TK]D-Fendercheckout time, later all
00:56.15*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
00:59.27drmessanoSo ummm
00:59.39drmessanoAbout that Exchange UM + Asterisk
00:59.51drmessanoAny way to see if TCP is working in Asterisk?
01:00.02drmessanoother than knowing tcpenable=yes?
01:00.09jayteegot two tcp capable phones?
01:00.32drmessanoWhat the hell is that gonna pro... oh, hang on..
01:00.45jayteeset them up in * with canreinvite=no
01:01.05drmessanoI got an ATA.. doing that now
01:01.12jayteethen test calling one from the other? hey! I'm just an idiot throwin shit out there. thought you knew that already
01:01.30*** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200)
01:02.06drmessanolol
01:02.20drmessanoIm gonna do the two ports independent
01:03.07jayteeyou mean one phone tcp and the other udp to make * do the transform?
01:04.11drmessanoUm.. oh um, Yeah, that's exactly what I was planning on doing before you mentioned that.....
01:04.29*** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-081-010.dsl.sil.at)
01:09.18drmessanoDamnit man
01:09.42*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
01:20.59*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
01:23.40*** join/#asterisk jer (n=jer@unaffiliated/jer)
01:33.32drmessanoI cant register Windows Messenger to Asterisk with TCP
01:33.39drmessanoIt works with UDP
01:33.58drmessanoUnless there's some other difference, TCP appears to be NOT working
01:35.05drmessanoOh
01:37.15drmessanoScratch that.. working now
01:37.15*** join/#asterisk kamanashisroy (n=kamanash@119.30.34.5)
01:37.16drmessanolol
01:44.04*** part/#asterisk korihor (n=korihor@200-71-160-1.genericrev.telcel.net.ve)
01:44.35drmessanoJaytee
01:45.31drmessanoAnyone using TCP with 1.6?
01:46.00etfonhomeyHow do you have Windows Messenger registering to Asterisk?
01:46.01mvanbaaknope ;)
01:46.40drmessanoI set the SIP options and click connect?
01:47.24etfonhomeyWhere in Windows Messenger do you have the ability to set SIP options.
01:47.26etfonhomey?
01:47.32drmessanoApparently I cannot make calls from UDP peers to TCP peers and vice versa through asterisk, which makes no sense
01:47.47drmessanoTools > Options > Accounts
01:48.20drmessanoand TCP to TCP isn't working.. almost like TCP is enabled but.. not so much
01:48.26etfonhomeyWhat version of Windows Messenger are you using?
01:48.38drmessano5.1
01:48.51drmessanoThe latest
01:48.57drmessanoor should I say, last
01:48.59etfonhomeyAh.  I have 4.7
01:49.06mvanbaakuse something decent
01:49.21etfonhomeyI do and it's not a Microsoft product.
01:49.23drmessanoDecent?
01:50.03mvanbaakyeah, something without an evil EULA
01:50.11drmessanoAh here we go
01:50.36mvanbaak;)
01:50.47coppicethe evil Eula should be a character in some epic tale of ancient Egypt
01:50.55drmessanoI actually came here for help with TCP in Asterisk, not a 4 hour MS bashing convo
01:50.58*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
01:51.24coppicedrmessano: they come free with every meaningful question
01:51.27mvanbaakhave you tried the tcp stuff with some other client ?
01:51.49mvanbaakor hardphone ?
01:51.59drmessanoI have neither
01:52.39drmessanoIm trying to get unified messaging working with Exchange.. and I wanted to make sure TCP was at least *working*.. I can register just fine
01:53.18mvanbaakwhat's the error on asterisk cli ?
01:53.34drmessanoCHANUNAVAIL
01:54.07mvanbaaktry a sip debug
01:54.08drmessanoI can call my IVR just fine with Windows Messenger using TCP
01:54.16drmessanoJust not another peer
01:54.49mvanbaakand exchange um cant use udp ?
01:55.00drmessanolol no
01:55.14drmessanoForgetting exchange UM
01:55.22drmessanoTCP device can reg and call IVR
01:55.35drmessanoBut not another phone thats using UDP
01:55.39*** part/#asterisk codefreeze-lap (n=murf@72.21.67.40)
01:55.46drmessanoSounds like something internal
01:56.15mvanbaakand this is true for non-messenger clients using tcp as well ?
01:56.36mvanbaaktry disabling reinvite
01:56.39drmessanoI have not tested any non messenger clients
01:56.46drmessanocanreinvite=no is set
01:56.55coppicedrmessano: until recently asterisk had a problem with an number of MS products. they tack "; charset=utf8" to the end of some of their SDP, in a perfectly valid way, but asterisk baulked.
01:57.11drmessanoOk, but its working with asterisk one way
01:57.15etfonhomeyGuess I need some kind of tcpenable in my sip.conf?
01:57.22drmessanoI can call my IVR
01:57.25*** join/#asterisk CunningPike (n=arodgers@204.239.8.157)
01:57.41mvanbaakcoppice: that's because asterisk does not support utf-8
01:57.51drmessanoI can call my wakeup call app
01:58.11coppiceno. its because asterisk was complaining about any trailing stuff
01:58.13drmessanoSo that leg is working 100%
01:58.23mvanbaakdrmessano: have you tried the other way around ?
01:58.35mvanbaakusing an udp device to call your tcp device ?
01:58.41drmessano.....
01:58.50drmessanoWe keep going over this.. let me lay it out again
01:59.06drmessanoTCP device calls IVR and internal apps - CHECK
01:59.16drmessanoTCP device calls UDP device - FAIL
01:59.23drmessanoUDP device calls TCP device - FAIL
01:59.26drmessanoTCP device calls TCP device - FAIL
02:00.17mvanbaakwhere TCP device == ms messenger
02:00.29drmessanoWhy should that make a difference?
02:01.10mvanbaakgheh
02:01.34drmessanoIsn't asterisk a B2BUA?
02:02.09mvanbaakwell, every client has their own special stuff
02:02.20drmessanoSure, but this client is working
02:02.21mvanbaaksome are compatible, some arent
02:02.36mvanbaakthat's why I want to know if this is true with other tcp clients as well
02:02.53drmessanoI dunno, I can run out and buy a polycom real quick
02:03.00mvanbaakthat way we know if it's a general tcp error or just incompatibility with ms messenger
02:03.07drmessano...
02:04.20mvanbaakanywayz, I'm going to bed
02:04.25mvanbaakSun Nov 16 03:04:02 CET 2008
02:04.28mvanbaaklatero all
02:05.38drmessanoWell, MS hate strikes again
02:05.58drmessanoObviously if it's an MS product involves, it MUST be at fault..
02:06.05drmessanoinvolved
02:07.27*** join/#asterisk l1quid- (n=liquid@pool-96-253-72-119.rcmdva.fios.verizon.net)
02:10.10*** join/#asterisk Daejeo (n=chatzill@118.219.208.186)
02:10.30Daejeoi want to trigger the call by email
02:10.55Daejeoany doc to get some idea?
02:11.12coppicedrmessano: who has said MS must be at fault, apart from you?
02:12.51drmessanomvanbaak actually made a couple comments implying such.. but no worries, trying eyebeam now
02:15.30drmessanoSame problem
02:15.48coppicehe suggested incompatibility. I don't think he actually said who's fault it might be :-)
02:16.05drmessanoeyebeam does the same using TCP
02:16.11drmessanosoooo
02:16.29raasdniltherefore... microsoft must be at fault :)
02:16.38drmessanoof course..
02:16.40raasdnilperfectly logical when you think about it :)
02:17.05raasdnilhas flash backs of monty python witch scenes
02:17.11drmessanoSo apparently TCP isn't completely working in 1.6
02:19.52coppicevarious people claim this to be true
02:20.08drmessanolovely
02:20.55drmessanoI guess there is a bright side
02:21.37jayteems messenger isn't SIP
02:21.42coppicenow you're delusional
02:21.58drmessanosure it is
02:22.23drmessanoWindows messenger is SIP and is actually a pretty decent softphone
02:22.37drmessanoTheres MUCH better
02:22.46drmessanoBut for a freebie
02:22.51drmessanoand for testing
02:23.02drmessanonot too bad
02:25.13coppiceif its considerably worse than the very best softphone it must be awful :-\
02:25.36drmessanoHA
02:25.59drmessanoTheres some halfway decent ones
02:26.06drmessanos/ones/one
02:26.25jayteeOffice Communicator supports SIP but the early versions of MS Messenger used it's own protocol. When did they add sip support to it?
02:26.32drmessano5.0
02:26.44drmessano5.1
02:26.54drmessanoLCS 2003 didn't have Office Communicator
02:27.07drmessanoIt's native client was Windows Messenger 5.0 and later 5.1
02:28.51drmessanoWindows Messenger 4.5 used Exchange Messaging, which was supported through 5.1
02:30.40*** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com)
02:35.17jayteewell, before I start testing 1.6 tcp with UM I'm going to do some packet captures of the SIP traffic between sipX and Exchange UM because I know that works at least so when I delve into the morass of *'s tcp implementation I've got something to compare it to.
02:35.22*** join/#asterisk MrNaz (n=mrnaz@210-84-62-81.dyn.iinet.net.au)
02:36.32*** join/#asterisk mateo_au (n=mateo@c122-106-221-182.belrs3.nsw.optusnet.com.au)
02:36.57drmessanoYeah well
02:37.07drmessanoyou got the ASS part right
02:37.12drmessano, pain in my
02:38.34drmessanoI simply cannot call a TCP device
02:38.59drmessanogrrr
02:39.18jayteetransport = tcp in the client's section of sip.conf?
02:39.31drmessanoyep yep
02:39.51drmessanoTrying on two different boxes here
02:39.59jayteefutures = porkbellies in commodity.conf?
02:40.02drmessanoIm about to get some alcohol
02:41.22drmessanoAt least I know now that its not exchange
02:46.05jayteehow'd ya figure that?
02:46.14drmessanoEyebeam doesn't work either
02:46.19drmessanoNeither does Messenger
02:46.22drmessanoSame symptoms
02:46.57jayteeyou mean testing to and from each other, not to Exchange
02:47.16drmessanoyes
02:48.12jayteemaybe that's why they didn't cover sip tcp in the Asterisk Advanced class even though we used 1.6
02:48.19drmessanolol
02:48.43drmessanoSee, the thing is
02:48.57drmessanoTCP client <> Asterisk works
02:49.07drmessanoTCP client <> Asterisk <> UDP/TCP Client does not
02:49.19drmessanoTheres something missing
02:49.36jayteethey're probably going to come out with an Asterisk Expert class that covers that. "Our Asterisk Expert class focuses on new features such as SIP TCP and TLS. By the end of the class if you've got SIP TCP and TLS working, then you're definitely an expert."
02:49.45drmessanolol
02:50.15drmessano5 hour lab.. "Configure a TCP device with asterisk"
02:50.20drmessano"And?"
02:50.23drmessano"No, thats it"
02:50.24jayteeso TCP to * to access something like Playback(boss-is-an-asshole) works?
02:50.34drmessanoyep yep
02:51.06drmessanoIts not umm.. trans--... transgendering
02:51.24jayteetransforming
02:51.25Maliutajaytee: of course you hacked it so the call is actually Payback(boss-is-an-asshole) :)
02:51.53drmessanotransportingporting
02:52.10drmessanoyeah, transforming.. Like a weak autobot
02:53.13*** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net)
02:53.17jayteeI'd love to find a new job and before I leave modify the dialplan so my boss's phone can only access a macro that plays a custom MOH, "Take this job and shove it!" by Johnny Paycheck
02:54.10MiccI can't believe all the problems I've been having was from a stupid sql trace file being too big.
02:55.16jayteeand if anyone calls him they'd get, "Warning: you are calling a grouchy idiot. If you're sure you wish to speak with him press 1 to continue, if you'd rather just skip talking to him and just leave a voice message press 2. If you've come to your senses and don't care for either just hangup."
02:56.50drmessanohttp://bugs.digium.com/view.php?id=13117 <-- tada
02:59.38jayteeso there's a patch then
03:01.03drmessanoA broken one
03:02.41*** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-010-032.mycingular.net)
03:03.03*** join/#asterisk dramman (n=Miranda@122.111.59.159)
03:05.44jayteedrmessano, check this one out too, http://bugs.digium.com/view.php?id=13523
03:08.54drammanI'm trying to set up a in/out sip trunk on asterisk.  I
03:08.56*** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net)
03:09.18*** join/#asterisk ManxPower (n=manxpowe@70.sub-70-223-194.myvzw.com)
03:09.43jkswhat is the difference between state "Ring" and "Ringing" for newstate events on the manager interface?
03:10.31drammanI've been led to believe that I need an [engin_out] and [engin_in] context, and in [genera] "register => 0212345678:pass@byo.engin.com.au/0212345678
03:10.39harry_vjks, what versio are you running?
03:10.53jksharry_v, 1.4.21
03:10.57harry_vgood
03:11.13jksharry_v, ?
03:11.15drammanOr should the register string be ..../engin_in  ?
03:12.45drammanIf I set it to .../40 it tries to route the call to SIP/40 when a new call comes in, but I actually want to be able to route it to a ring group (with voice mail fail-overs etc)
03:17.16baliktadyour register string looks OK as is
03:17.32baliktadyou need an incoming context in your dialplan for the call to be routed to
03:17.40baliktadthen you can handle the call however you see fit
03:19.11drammanCan you see anything wierd/wrong in this? http://pastebin.com/d55c8876
03:21.17drammanFor incoming calls, it should be going to [DID_engin_36_in]
03:21.17baliktadso right now your incoming calls from engin get answered and then just ring extension 7000
03:22.24baliktadnot sure why line 209 has extension s with a period after it
03:22.59drammanno - "chan_sip.c:13885 handle_request_invite: Call from '' to extension '0290115436' rejected because extension not found."
03:23.21baliktadpaste a sip debug
03:25.11drammanhttp://pastebin.com/d1b43d037
03:26.00drammanI've actually commented out line 209
03:26.19baliktadline 43 - your incoming call isn't getting matched to a sip peer
03:26.25baliktadso the call is going to your default context
03:26.36baliktadwhich of course doesn't have an extension matching the incoming number
03:27.10drammanshouldn't it match [engin_36_in] at line 49?
03:27.53dramman(line 49 of the first (config) pastebin)
03:27.54baliktadline 54, you have host = byo.engin.com.au
03:28.15baliktadbut the call is coming from 203.161.164.69
03:28.35baliktadargh, same
03:30.39drammanI'm confused as to why I need to define a register string _and_ [engin_36_in] in sip.conf (and how to get them to marry-up)
03:31.14baliktadthe register string tells * to continually register with your ITSP, so it knows where to send calls to
03:31.17*** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer)
03:31.46baliktadthe [exgin_36_in] context provides all the details about handling a call that comes in
03:31.55drammanis [engin_36_in] a sip extension, or a context?
03:32.23baliktads/context/account
03:32.33baliktadif it's in sip.conf, it's a SIP account
03:32.38baliktadyou only have contexts in your dialplan
03:32.57drammandialplan == extensions.conf?
03:33.02baliktadyes
03:33.14drmessano[Digg] Concept: Taking Social Networking To The Next Step  <--- Face to face?
03:33.51jayteeNoooooo!!! I don't want to leave my house!
03:33.52drammanso what's the purpose of the last bit of register string after "/"?
03:33.56drmessanoHAW
03:33.57baliktadhave you reloaded your sip module and dialplan since making the changes you have pastebin'd?
03:34.15drmessanoWho the hell wants to interact with live people?
03:34.28baliktadthe last part tells asterisk what extension it should try and match when the call makes it to your dialplan
03:34.29drmessanowrites off the whole idea
03:34.30drammanyes, continually doing "module reload"
03:34.37jayteeI've changed my mind, when I grow up I want to be a choo-choo instead of a firetruck
03:35.02drmessanolol
03:35.24drmessanoI want a copy of Microsoft PBX Simulator 2007
03:35.32jayteeor maybe a zamboni
03:35.38jayteelol
03:35.40Maliutajaytee: too much Vanilla Ice? decided you want to be a train and he'll be the caboose? ;)
03:35.52drmessanoOMG
03:35.55drammanSo by saying /0290115436 it'll call "extensions.conf::0290115436" which is essentially the same as [DID_engin_36_in]?
03:36.07drmessanoNot so much the Vanilla Ice bust out.. but the reference
03:36.25drmessanoMaliuta: The person using a Vanilla Ice reference on someone can't win
03:36.26baliktadfirst you need to get the call to match up to a SIP account
03:36.34Maliutais frightened that he remembers VI lyrics
03:36.36baliktadit has to be matched to a sip peer for asterisk to accept the call
03:36.39jayteedrmessano, no one's gonna slide up to my bumper if I have anything to say about it
03:36.48baliktadonce it does, it will find the context= line for that sip peer
03:37.03drmessanoThat's like "HA, you're just like that ABBA song"
03:37.03Maliutadrmessano: I win by default
03:37.07drmessano^ Fail
03:37.16drammanso the problem lies in "sip.conf::[engin_36_in]"?
03:37.20baliktadthen it looks up that context in your dialplan (extensions.conf) and searches for extension 0290115436
03:37.23*** join/#asterisk rcy`` (n=rcy@S01060002553240a8.vc.shawcable.net)
03:37.24Maliutadramman: but _you're_ the dancing queen in here ;)
03:37.38drmessanotab FAIL
03:37.39baliktadyes, there is a problem matching to your sip peer
03:37.42*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
03:37.52Maliutas/dramman/drmessano/
03:37.52baliktadfor some reason I can't yet figure out
03:38.07drmessanos/tab fail/HA!
03:38.24drmessanos/*/*
03:38.32drmessanoomg a loop
03:38.34Maliutareminds drmessano that he wins because he is "The Illegitimate Son of God"
03:38.39jayteethey taught us in class to make a sip peer account for our SIP provider and a sip user account to match
03:39.58jayteeMaliuta, who's the Illegitimate Son of God? you or drmessano?
03:40.20Maliutajaytee: well obviously it can't be him
03:41.02drmessanoDad says to STFU
03:41.12jayteereminds me of right before the election, I was getting all kinds of junk mail and fearmonger calls from the RNC slander/fearmonger unit.
03:41.42MaliutaKMFDM says "Spit Sperm"
03:42.25*** join/#asterisk stencil (n=stencil@unaffiliated/stencil)
03:42.33jaytee"Did you know Barack Obama is the Anti-Christ?" "No! I didn't but now that I do he certainly has my vote. I was leaning towards McCain/Palin because you just know that bitch is evil. Thanks for pointing that out! Have a good nite and Hail Satan!" Hangup()
03:42.44*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
03:43.16Maliutajaytee: who did you get to record that for your autodialer?
03:43.45stencilYa, but Obama gives us cheap gas!
03:43.52drmessanoOh it was funny
03:44.03jayteeAllison wouldn't do it so I hired a dancer from P.T.'s Gentleman's Club
03:44.08drmessanoI was reading about Paul Broun calling Obama a socialist, etc
03:44.12baliktaddramman any luck?
03:44.19*** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net)
03:44.20drammanjaytee: : [engin_36_out] is my peer, [engin_36_in] is my user
03:44.27drmessanoHes this fucker from GA that kept sending us spam for WEEKS
03:44.48drmessanoi am gonna e-mail him and ask for his resignation
03:46.41harry_vas long as obama has the right team then it will work.
03:46.45Maliutathat give Socialists a bad name
03:47.05jayteedramman, then you need to put context="nameofcontext_to_handle_incoming_call" in your [engin_36_in] account
03:47.31baliktad...he has that, line 52
03:48.28jayteesupposedly in 1.6.2 users and friends go bye-bye and everyone becomes a peer. Part of the new equality guaranteed by the incoming administration.
03:48.42baliktadhis incoming call isn't getting matched to any SIP account
03:48.47baliktadhahah brilliant
03:49.02jayteethe first part of that is true, the rest was humor
03:49.11drammanjaytee:  yes, I have "context = DID_engin_36_in"
03:49.19jayteeat least that's what they said in class this week
03:49.56jayteeand is the context actually named DID_engin_36_in ?
03:50.20drmessanojaytee: You are my peer.. my base
03:50.21drammanyes
03:50.43jayteedrmessano, and you are the wind beneath my wings ;-)
03:50.47harry_vis there no zap show chanells in core?
03:50.52drmessanocovers his butt
03:50.57jayteelol
03:51.07Maliutadrmessano: here comes that train again
03:51.23Maliuta*cough*Ice Ice*cough*
03:51.45jayteeharry_v, if you type help zap at the CLI what do you see?
03:51.47drammanshould I set nat, registersip...?
03:51.50drmessanojaytee: If you think about it.. Sounds like peers/users are going away and "friend" is becoming "peer"
03:52.09jayteedrmessano, that's kinda what I was thinking
03:52.18harry_vno such command
03:52.22drammanmy outgoing calls are working
03:52.34drmessanoA little sed and internal rewrite of friend to peer for deprecation
03:52.36Maliutaharry_v: have you loaded chan_zap?
03:52.43jayteeharry_v, did you compile your * install?
03:52.44harry_vcompiled
03:52.49harry_vyes
03:52.53jayteedid you compile zaptel first?
03:52.57harry_vno
03:53.00drmessano......
03:53.03drmessanobingo!
03:53.06jayteeah, wrong order pal
03:53.06harry_vhehe
03:53.15jayteelibpri, zaptel, asterisk
03:53.16drmessanoYou get to drink....
03:53.18drmessanoFROM THE FIREHOSE
03:53.18harry_vohh gee, 3 years since last time i compiled it
03:53.23baliktaddramman can you paste the output of sip show peers
03:53.45harry_vso basicly recompile zap then asterisk
03:53.50drmessanoharry_v: $5 fine for whining
03:54.18harry_vno its called recall everything from five years ago :)
03:54.20jayteeharry_v, just go into the zaptel source directory and type make config, then go into the asterisk source directory and do a make&&make install
03:54.29*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
03:54.36harry_vI know jaytee :)
03:54.48jayteenote the double ampersand there, or you could break it out into separate commands
03:54.54drammanhttp://pastebin.com/d68724f7e
03:55.00jayteeactually might want to do a make clean first
03:55.20baliktadok, it's as I expected
03:55.45jayteeengin_36_in isn't registering
03:56.03baliktadyour engin_36_in isn't being processed as a peer
03:56.44*** part/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net)
03:56.56drammanit's set to "type=user"
03:57.23baliktadif you are making calls in and out to this provider, it's probably easiest to just combine your engin_36_in and engin_36_out accounts into one engin_36 account of type friend
03:57.47harry_vright now though, my real issue is vm not kicking in the end of the 25 second ring duration.
03:58.13baliktaddoh
03:58.37baliktadI always get confused too
03:58.46harry_vohh wow, for some off reason its adding a module
03:59.18harry_vprobebly do do with the zaptel needing sounds.
04:00.06baliktadif you want to accept calls from someone, you need to have an account as type peer or friend
04:01.00drammanI thought peer was for making calls
04:01.22drammanHave I got peer/user swapped?
04:02.12drammanI've not been able to find documentation which makes it very clear - it usually just lists the options and assumes that you know
04:02.23jayteedramman
04:02.30jayteedid you look in the book?
04:02.40jaytee~book
04:02.41jbothmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
04:03.32harry_vjaytee, thanks for the reminder. btw, what version are you running?
04:03.36Maliutawe should train him to respond to ~worshiphere
04:03.52*** join/#asterisk ManxPower (n=manxpowe@55.sub-75-202-121.myvzw.com)
04:06.01ManxPowerregistration notifies the remote server what ip address is associated with that userid/password and request what extension the calls be sent to.  It does NOTHING else.
04:06.09jayteeharry_v, depends which server. I've got 1.4.15 and 1.4.22 in production and 1.6.0.2 in test
04:06.58harry_vManx, am i correct that 1.4.X did away with n+101 for the number sequence in a dial pattern for extentions.conf?
04:09.31jayteen+101 was priority jumping, not dialed number manipulation
04:09.41harry_vyes
04:09.51harry_vthat was what i was refering to
04:10.11ManxPowerharry_v: I assume so but the official document is called...can you guess?
04:10.17jayteeit still works in 1.4 but it was listed as deprecated and you should use labeled priorities instead.
04:10.32jayteeNOBODYREADSME.TXT
04:10.38ManxPowerjaytee: nope.
04:10.45harry_vI did read it in a document. I am trying to troubleshoot my strange voicemail issue and thought this may have had something to do with it.
04:10.46jayteeUPGRADE.TXT?
04:11.03jayteeor UPGRADE1.2-1.4.txt
04:11.06ManxPowerUPGRADE.txt AND UPGRADE-1.2.txt both included in the source.
04:11.11*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-219-34.phlapa.east.verizon.net)
04:11.48jayteeI think I've seen those files in a folder somewhere. wonder what's in 'em? :-)
04:12.05ManxPowerjaytee: nothing important, I'm sure!
04:12.38ManxPowerI can see how someone using 1.4 might think they don't need to read UPGRADE-1.2.txt, but the 1.2 info is NOT duplicated in the 1.4 info, so you need to read all of them
04:13.06*** join/#asterisk kisu (n=hexago@daniel1117.broker.freenet6.net)
04:14.30harry_vi will have to troubelshoot this latter.
04:14.34jayteeManxPower, there are certain things in life that instill in some of us to read the readme.txt file. Like spending a couple hours trying to get the scsi driver loaded on a Netware server running on a PS/2 Model 90. Then you find a little cryptic blurb in the readme that says. "You must have slot=99 in your load line for the embedded scsi controller on the Model 90 to initialize properly.
04:15.01ManxPowerjaytee: these youngsters don't know how good they have it!
04:16.07jayteenope, and I still resist myself. I start thinking, hell no I don't want to read that! I don't want to go home at a normal hour, I want to work late and eat crap snack food instead of a nice dinner and stare at this monitor till my eyes start to bleed and feel like they've been sandblasted.
04:16.10ManxPowerjaytee: Some people think Asterisk does not have good docs -- try the Novatel Wireless (CDMA) Toolkit and you'll see what bad docs are.
04:17.02ManxPowerThey actually FORGOT to include the sample source code that is referenced all over the SDK in the Linux build of the SDK.
04:17.21jayteeManxPower, my Digium backpack had a copy of AsteriskNOW for Dummies that comes with the software on DVD. Must be a consolation prize :-)
04:17.47Maliutathey're giving you coasters?
04:18.03ManxPowerAsteriskNOW for Dummies?  Isn't that like "Eating Celery for Dummies"?
04:18.18jayteeMaliuta, no. Coasters come from AOL
04:18.37Maliutaor encouraging you to engage in team work and build a barometer?
04:18.48jayteeCelery, lol. that's my pet name for the Intel Celeron processor
04:18.50*** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net)
04:20.13Maliutajaytee: you never got exposed to MSDCD's?
04:20.22drmessano1.6.0.2 is out?
04:20.28MaliutaI had literally close to a thousand
04:20.43Maliutadrmessano: it might be in the closet
04:20.55drmessanocloset?
04:20.57drmessanooh
04:21.01drmessano......
04:21.05Maliutadrmessano: I haven't seen an announcement, but I didn't see the 1.6 announce
04:21.50*** join/#asterisk sah-work (n=Bawbatos@12.14.133.181)
04:22.24jayteeoh, sorry. that was a typo. 1.6.0.1
04:22.25drammanI've commented out [engin_36_in] and am trying to do everything with [engin_36_out] (will rename later if it works)
04:22.36drammansip debug still says "
04:22.59drammanno matching peer or user for '203.161.160.69:5060'
04:25.11dramman(whoops, [engin_36_out] had "hassip=no") - now saying:
04:25.23*** join/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com)
04:25.24drammanNOTICE[3967]: chan_sip.c:13885 handle_request_invite: Call from '0290115436' to extension 'DID_engin_36_in' rejected because extension not found.
04:27.15jayteehow does someone dial DID_engin_36_in from a phone? they didn't teach us that in class.
04:27.18Maliutadramman: next question do you have an exten => 0290115436 in that context
04:27.49ManxPowerjaytee: his register statemnet is prolly screwed up
04:28.05Maliutajaytee: how do you dial SIP/nikolai@voip.kissmyass.com from a phone?
04:28.35Maliuta:)
04:28.56drammanIn extensions.conf::[DID_engin_36_in] I've got "exten => s,1,NoOp(Call from Engin)" "exten => s,n,Dial(SIP/7000,20)
04:29.28ManxPowerdramman: "s" means "the technology is too stupid to send us a dialed number", usually FXO ports.
04:29.37Maliutait's not going to 's'
04:30.02MaliutaManxPower: no, my ITSP sends to s
04:30.16ManxPowerMaliuta: then your ITSP is too stupid.
04:30.29dramman...now, there isn't actually a [7000] defined in sip.conf, but there is a "exten => 7000,1,Goto()" in extensions.conf::[default]
04:30.37MaliutaI guess they expect you'll have just on device or on context per account
04:30.50drammanIsn't "s"
04:30.54drammaneverything?
04:31.08ManxPowerno, "s" means "NOTHING"
04:31.17ManxPower"_." means "everything"
04:31.43jayteeor s means "um, I guess I'd better start here since they didn't give a number to dial"
04:31.50drmessanos = stupid  _. = "No period, oh shit"
04:32.33ManxPowerI assume ATFOT talks about "s"
04:32.55jayteeManxPower, careful with the assume, buddy!
04:33.14MaliutaManxPower: IIRC that's a diaplan for and FXO
04:34.34jayteewow, check this out!!! it's awesome!!! http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
04:34.57ManxPowerIf your DID is 5045551212 then usually you would exten => 5045551212,1,Dial(whatever
04:35.20drmessanoYou know what happens when you assume
04:35.26drmessanoYou make an ass out of Maliuta
04:35.33drmessanoducks
04:35.51Maliutakicks drmessanos ducks in a row for him
04:36.32drmessanothrows a handful of fair dinkum
04:36.56*** join/#asterisk hadronzoo (n=user@gateway.publicvpn.net)
04:37.35Maliutakicks drmessano innanuts
04:38.03drmessanoyep, you got me there
04:38.11ManxPowerjaytee: what did you think of the class?
04:38.15jayteeman, this part of Huntsville is boring
04:38.28drmessanoWhen do you go back?
04:38.35drammanOk, I've swapped the "s" with "_." and it's trying to dial SIP/7000 - not exactly working, but a lot better
04:38.46jayteeManxPower the class was awesome, right up till the point where I ran out of time doing the practical lab and scored a 45
04:38.57jayteeI fly out tomorrow
04:39.12ManxPowerdramman: unfortunately you cannot just take bits and pieces and expect it to work as expected.  Go read the Asterisk Book.
04:39.38ManxPowerbecause with _. you will have dialplan loops as the same extension will match 12345 and match "h" and "i"
04:39.50drmessanoI tried to cut and paste out of the book and now my monitor wont shut off.. damn Elmers glue
04:39.55ManxPowerSo go read the book and save yourself weeks of problems.
04:40.16jayteepersonally I think 90 minutes for a lab to compile, install and configure 3 phones with voicemail and an IVR and a SIP provider a bit too tight. If I'd had 2 hrs I would have passed.
04:40.32ManxPowerYou almost NEVER want to "match all"  In fact the CLI should complain if you use _.
04:40.33MaliutaGo read the book and save the regulars from hunting you down and beating you to death with it
04:40.52drmessanojaytee: Where you screwed up is you forgot your years of watching 80s TV shows
04:41.24ManxPowerjaytee: Did outside/voip calls sound crappy?
04:41.37drmessanojaytee: You should have downloaded AsteriskNOW, spent 10 minutes configging the box and the phones, and then adjusted your collar and ate some pizza, with a smirk on your face
04:41.40jayteewell, I had the option to not use and configure my T1 card but I foolishly thought "what the hell, I've done this before"
04:42.10Maliutais thankful the tv show he is watching right now is '70s not '80s
04:42.12drmessanoBecause in the 80s, you would have gotten away with it
04:42.22jayteeManxPower, in the class we were just emulating outside calls, not actually going outside
04:42.45drmessanoand your teacher with the unusually large glasses would have had to accept it and look stupid
04:42.48ManxPowerin the fast start we actually made some outside calls.
04:43.03ManxPowersounded like trying to send audio thru a broadcast storm
04:43.05jayteedrmessano, Jared doesn't wear glasses
04:43.26drmessanojaytee: He would if this was growing pains or Family Ties
04:43.31jayteelol
04:43.41drmessanoBUT
04:44.08drmessanoAfter all that coolness, you would have had to confront skippy or boner about their drug problem
04:44.09ManxPowerI've spent the past 2 days actually setting up a network in my cabin instead of only thinking about setting up a network in my cabin.
04:44.24jayteethis was only the 3rd Advanced class, they're still fine tuning the curriculum
04:44.39ManxPower*nod*
04:44.55Maliutajaytee: anyone interesting running them?
04:45.11drmessanojaytee: to put that into context here, after "cooly outdoing" Jared, you would have to talk to Russell about his blackberry addiction
04:45.13jayteeMaliuta, running what?
04:45.14drammanMy config files are basically taken from the asterisk samples and the book.  On page 127 it defines [incoming] exten => s,1,Answer()
04:45.29Maliutajaytee: the classes
04:45.38drmessanoTo which Russell would have replied "I dont have a problem dude, dont be downer"
04:45.41drmessanoTo which Russell would have replied "I dont have a problem dude, dont be A downer"
04:45.49ManxPowerdramman: The asterisk .sample files are designed to show as many options as possible, not to actually work.
04:45.53drmessanowatched too much 80s TV
04:46.09jayteeMaliuta, my class was taught by Jared Smith which if you look on the front page of the "The book" you'll see his name along with two others
04:46.22Maliutapasses drmessano a pile of CHiPS episodes to watch
04:46.26drmessanoOh god
04:46.31drmessanoNow Jaytee is gonna be a name dropper
04:46.38drmessanoNice knowing you, slick
04:46.58jayteedrmessano, he asked. I didn't volunteer it
04:47.08drmessanoIm just screwing with ya
04:47.21MaliutaI can try an out drop him, but I think mine are fairly run-of-the-mill names
04:47.28drmessanoBesides, you didnt elaborate enough for a full, egomaniacal namedrop
04:47.28jayteebut when we were having lunch with Mark...... :-)
04:47.41MaliutaSuter was there?
04:47.42drmessanoLet me fix your sentence
04:47.56jayteewho's Suter?
04:48.06MaliutaI haven't seen Mark Suter in ages, I thought he was still in Canberra
04:48.16Maliutataps his nose
04:48.48jayteenever heard of him. is he a notable name in the * "community"?
04:49.11Maliutano, but he is a notable name in these parts
04:49.23Maliutagoogle is your friend
04:49.31drmessano"Maliuta, my class was taught by Jared Smith, which if you look on the front page of the "The book" you'll see his name along with two others, who I knew back from the days of installing key systems for trademark telecom back when we were soldering frames and praying for flashlight batteries :)"
04:49.33drmessanoThere
04:49.37drmessanoThats about right
04:49.39jayteethrows Maliuta a Scooby Snack
04:50.45jayteewiseass! :-)
04:51.37drmessano"I have been using Asterisk since 0.0.9.  Notice the earliest posted on the download site is 0.1.0?  ;)"
04:51.43drmessanoI think thats FTW
04:52.02Maliutaback when I was trading patches with W Richard Stevens .....
04:52.21drmessanoOh god, that reminds me of this time stallman and I...
04:52.43jayteeI got to meet Qwell and russellb in person which to me at least is a nice thing to put a face with a nick in IRC. I'd like to meet drmessano but I know his Facebook pic is a fake and he's really a bot setup by Kerry Garrison
04:53.27MaliutaI actually sent him some patches (for a problem they had already solved with the d/l code for unix network programming). Nice guy, we traded email for about 2 weeks
04:53.27drmessanoNo, I am really Kerry Garrison
04:53.36drmessanoNow kiss my ass.. and give me back my camera
04:53.45jayteeMaliuta, who? russell?
04:55.31filewobbles
04:55.33jayteeboth he and Qwell are really nice people and very very smart. russell said he doesn't do dialplan work so he hangs in here on occassion to get a feel for how we're all using it.
04:55.53jayteeweebles wobble but they don't fall down!
04:55.55Qwelljaytee: developers don't actually USE the stuff
04:55.56Maliutajaytee: W Richard Stevens
04:56.04fileI use it!
04:56.45jayteeyes, you do!
04:56.54jayteeand I for one am grateful for that
04:57.02ManxPowerI hink my TiVo is freaking out.
04:57.05drmessanoQwell: You dont need to tell us the developers dont use this effing thing.. The code wreaks of it.
04:57.15drmessanoducks
04:57.20jayteefile, and thanks for the tip on preloading grammars by the way
04:57.36filejaytee: working well?
04:57.47*** join/#asterisk styelz (n=yoohoo@egg.vividas.com)
04:57.59jayteeQwell, ignore him. Kerry compiled the extra-snarky module when he created the drmessano bot :-)
04:58.37jayteefile, yes. each section of the IVR no longer has a slight lag like it did.
04:58.57filejaytee: good
04:59.18drmessanoI mean, just look at the config files.. The damn config opens bear the lingering stench of "we're not end users, please enjoy crypticnoncryptic=yesnoyes as our non-confusing treat"
04:59.24drmessanoBah humbug
04:59.53drmessanoheh
04:59.53jayteeI was working on it over VPN today and adding some other tweaks to it I picked up from the class
05:00.06drmessanoActually, I owe file a beer
05:00.16fileI don't normally drink beer
05:00.31jayteewhat's your favorite poison?
05:01.08drmessanoI got G726 running with one of my providers.. had to use g726aal2, and actually understood all the cryptic BS about G726 thanks to him
05:01.09filepurple haze martini, or a good strawberry daiquiri
05:01.10*** join/#asterisk Kdas (n=Kdas@c-98-207-95-143.hsd1.ca.comcast.net)
05:01.16Kdaswass up ????
05:01.16drmessanoWorked beautifully
05:01.46Kdasanyone know why voicemails on my asterisk box don't alert my handytone286?
05:01.53drmessanoGenetics
05:02.09jayteewhat's in a purple haze? I've had a pomegranate vodka martini and it was tasty
05:02.25jayteehahaha, genetics
05:02.42filejaytee: pretty much that
05:03.08jayteeah, never heard it called a purple haze. you're in Toronto, right?
05:03.16fileMoncton, NB
05:03.43drmessanoPretty much the answer to "Why doesn't my grandstream ______" is "It's a grandstream"
05:03.44jayteewow, up on the far northeastern tip of the continent then.
05:04.16filejaytee: yup
05:04.21jaytee"I can see Greenland from my house!"
05:04.55drmessanoIts a little known fact that Grandstreams are based on later models of ColecoVision game consoles and use old TV color burst crystals for their PLL
05:05.13drmessanoIf you listen closely during a call, you can hear pole position
05:05.14jayteehahahaaha
05:05.23jayteerofl
05:05.38Kdasso... anyone know why i am not geting voicemail alerts on my ht286 ?
05:05.49drmessano"because it's a grandstream"
05:05.54drmessano~grandstream
05:05.54jbotgrandstream is, like, the Yugo of VoIP hardware.  Run.  Run away now.
05:06.00*** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net)
05:06.05echelonhi! :)
05:06.08Kdas:((
05:06.12echelonwhat's a good ATA?
05:06.16coppicegrandstream are one the very few makers to write all their own software, and the basic software isn't at all bad. its their QA that sucks
05:06.17echelonfor regular phone
05:06.19drmessanonot a grandstream
05:06.28jayteelinksys PAP2-T
05:06.45echelonhmm.. i've been hearing that as well
05:06.54echelonit supports fax?
05:06.56Kdasso i am screwed ?
05:06.57jayteeor if you want a little more bang for your buck, SPA2102
05:07.09coppicePAP2T == good ATA if you don't use FAX
05:07.11coppiceSPA2102 == good ATA if you do use FAX
05:07.27echelonit has FXO though?
05:07.35drmessanoNo
05:07.41drmessanoWho said FXO?
05:07.56coppiceSPA3102 is like an SPA2102 but one of the ports is changed to FXO
05:08.29jayteeyeah, the 3102 has a router and FXO
05:08.37jayteeand an FXS port
05:08.48jayteethe 2102 is just 2 FXS ports
05:09.35jayteeKdas, do you have a mailbox= statement in the sip.conf for that user?
05:09.44Kdasjaytee, yes sir
05:09.52echelonoi.. $63
05:10.21jayteelose the sir, that's like putting a chandelier in an outhouse
05:11.04jayteeKdas, been ages since I setup an HT286
05:11.19jayteecan't remember if there were any MWI options.
05:11.24Kdasjaytee, mwi ?
05:11.56*** join/#asterisk Vco (n=Vco@S0106000db912f754.cg.shawcable.net)
05:12.07jayteeMessage Waiting Indication. On a real phone it would make a lamp blink until you listened to the voicemail. On an analog phone on most systems it gives stutter dialtone
05:12.18Kdasjaytee, oh wait a second i have a mailbox= on my [callwithus] not under my [phone]
05:12.40Kdasjaytee, yea ht286 subosibly supports that
05:13.54jayteeKdas, you need to add the mailbox="mailbox#"@default (or some other context if you've setup different ones in voicemail.conf
05:14.27Kdasjaytee, do they go under the phone or my voip provider contexT?
05:14.37jayteein the sip.conf for the sip user you've defined for the FXS port on the Handytone. Asterisk doesn't know it's an FXS since it's an ATA.
05:15.11Kdassorry i didn't understand that
05:15.12jayteeKdas, the phone
05:15.24Kdasjaytee, ok i set that i will test that
05:15.39jayteein your sip.conf file you have a user set to the ATA, correct? because it's SIP to the ATA.
05:16.17Kdasjaytee, yes sir
05:16.27jayteeyour handydandytone registers to * as a sip peer.
05:17.01jayteeand you setup a user account in sip.conf for that peer and that's where you set the mailbox= statement.
05:17.09jayteeKdas, have you read the book much?
05:17.12jaytee~book
05:17.13jbotwell, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
05:17.41Kdasjaytee, yea i just been gone for a while till i got sick of not geting a voicemail flash
05:18.26Kdasjaytee, ok i get it so my handytone is blinking but my phone dosent seem to be getting it
05:19.13jayteeuse the web interface on the handydandytone to check if there's a stutter dialtone option for MWI
05:21.28Kdasok well right now i think its updating firmware but thatnks for help ;)
05:22.14jayteeman, I wish tomorrow I'd get on my flight and after 30 minutes in the air they'd come over the intercom and say, "due to severe weather and a backup at O'Hare we will be forced to land at Indianapolis for a layover until the weather clears." But then they probably wouldn't let me get my luggage.
05:22.35Kdashaha
05:23.29jayteeI can get direct flights to just about anywhere from Indianapolis except for Huntsville.
05:26.22jayteeI hate layovers at O'Hare. I always feel like Tom Hanks in that movie where he had to live at the airport
05:26.51harry_vhehe
05:27.07harry_vThat was a pretty funny movie.
05:29.27harry_vwell, its not as bad as taking a military HOP
05:29.39harry_vBut at least it is a free ride.
05:30.56drmessanoYour in Canada, right?
05:31.01drmessanoyou're
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05:32.16jayteewho's in Canada?
05:32.21drmessanoharry_v
05:32.24jayteeah
05:32.29drmessanoHe said "Military"
05:32.39drmessanoI was confused.. didn't know canada had one
05:32.55jayteenever heard of the RAF?
05:33.21drmessanoIs that like the RCMP?
05:33.24jayteeI'm trying to remember their air force stunt team's name
05:33.50jayteelike the Blue Angels or the USAF Thunderbirds. They put on a hell of a show.
05:33.58drmessanoI seriously doubt canada has an air force.. Where the hell would they put the horse...
05:33.59harry_vyes
05:34.15harry_vdr, im a US resident living up here.
05:34.29drmessanoIn Vancouver?
05:34.42harry_vCanada has a Airfoce. Every heard of Canadian F-18's?
05:34.52harry_vIn the Vancouver area
05:34.55jayteedrmessano, hehe, I'm former USAF and I worked with some Canadian RAF people in the past and they were top notch professionals.
05:34.56drmessanoOMG if you see Richard Dean Anderson, tell him I SO LOVE MACGYVER ZOMG 4EVA
05:35.13jayteerofl
05:35.21harry_vjaytee, what was your AFSC?
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05:35.35harry_vyea, he is cool
05:35.36jayteeharry_v, 30750
05:35.41drmessanoI stopped flying jets after I got shot down over Kumbang
05:35.50harry_v:)
05:36.20harry_vI worked on the Jollies
05:36.21jayteehe still comes in here once in awhile looking for drmessano
05:36.35Juggiejaytee, blue birds
05:37.03drmessanoI was 20 clicks north of Hunglad and was taking heavy fire from Tong-Po and most of General Tso's army
05:37.06harry_vGone are the days of working on the flightline and dealing with weather conditions
05:37.09jayteeJuggie, the precision flying team? that sounds right
05:37.28harry_vI imagine nam was pretty bad
05:37.32jayteeJollies? the Jolly Greens? those big ass dual rotor helos?
05:37.35harry_vyes
05:37.43harry_v3/53's
05:37.48Juggiejaytee, that is right :)
05:37.52harry_vdual=chinook
05:37.53jayteeah, in the USAF? or another branch?
05:37.57harry_vusaf
05:38.01harry_vSpecialOps
05:38.06jaytee:-)
05:38.11drmessanoharry_v: That wasn't vietnam, that was the buffet line at the Ming-Yat restaurant.. but almost as traumatic
05:38.27jayteeharry_v, Pararescue?
05:38.32harry_vrotorhead
05:38.43drmessanorotorhead?
05:38.47drmessanoPropellorhead
05:38.49harry_vcrew chief
05:39.07Juggieah
05:39.14jayteemy dad was a crew chief on B-17's in the 8th in England during WWII
05:39.15Juggiejaytee, i was close
05:39.17Juggiebut not right
05:39.18Juggiesnowbirds
05:39.21Juggienot bluebirds
05:39.29harry_vThat was how i started my career. Still have a strong interest in anything mechanical.
05:39.30drmessanoMy dad was kicked out of Vietnam
05:39.31jayteeJuggie, YES!!! that's it. thanks dude
05:39.37drmessanoThats his claim to fame
05:39.41Juggiehttp://www.snowbirds.dnd.ca/site/index_e.asp
05:39.55harry_vIf I had the money and a shop would not mind buying a T-58-GE-5 Turbo shaft jet engine.
05:40.03harry_vOr some other variant
05:40.08jayteeharry_v, my dad's nickname growing up was Tinker. he loved taking things apart and putting them back together.
05:40.23jayteehe was an excellent mechanic
05:40.25drmessanoMy dad annoyed the US Marine corps so much, and as so insubordinate that instead of letting him run in front of bullets and die, they chose to kick him out of the country
05:40.30harry_v:) I was one of those. But money was a issue.
05:40.39harry_vhehe
05:40.40jayteeand guess what base I got stationed at?
05:40.50harry_vReklavic?
05:40.52harry_v:)
05:41.06harry_vor some horrid base on the Alutian Chains
05:41.07jayteeno, Tinker AFB in Oklahoma City
05:41.12harry_vohh
05:41.19harry_vman that area is flat
05:41.21jayteeyep
05:41.34harry_vI was at Sheppard tx so I know :)
05:41.49harry_vkinda strange to see a city 20 miles away.
05:41.49jayteeI've been to Cheyenne Mountain in Colorado Springs and Elmendorf AFB in Alaska
05:41.55harry_vnifty
05:42.11harry_vhow many doors do thay have in that mountain?
05:42.16drmessanoI remember when I was at Cheyenne mountain
05:42.23drmessanoPlaying tic tac toe with the WOPR
05:42.26drmessanoGood times
05:42.36harry_vyour kidding right?
05:42.39jayteeI didn't count em. I saw about 4 huge blast doors
05:43.05drmessanowell, they called me in because they had a problem
05:43.16harry_vi see
05:43.17jayteebut there might have been more. I only got to go to certain areas inside.
05:43.24harry_vright
05:43.26drmessanoSee, the computer thought it was playing Globalthermonuclear War
05:43.37drmessanoBut it was a simulation
05:44.10drmessanoSo I had to go in and play a hellacious round of tic tac toe to teach it that theres more to life than who wins or loses
05:44.17harry_vI thought this was of movie making game playing. your kidding right?
05:44.45jayteemy favorite movie about computers is Colossus: The Forbin Project. The idea of the programming code for an AI that takes over the world being written and input via punchcards always cracks me up.
05:44.59harry_vmy base was perhaps one of two that stored all the nucks.
05:45.06harry_vBut i wont say where it was :)
05:45.31drmessanoWell, the WOPR had a few bugs
05:45.43drmessanoI blame Mr McKittrick
05:45.46harry_vI really enjoyed the movie
05:45.53jayteeharry_v, I bet i could guess :-)
05:45.55drmessanoHe looked a lot like Dabney Coleman
05:46.37drmessanoAfter the computer guessed the launch codes, man, I thought we were toast
05:46.44harry_v:)
05:46.47drmessanoCPE1704TKS it read
05:46.52jayteeharry_v, New Mexico?
05:46.56drmessanoTalk about jaws dropping
05:47.34drmessanoBut man, when that guy from Alaska was like "Sir, um sir, we're still here"
05:47.42drmessanoThat... was win
05:47.55harry_vI thought that was great
05:48.21harry_vsuch a good movie of its time. Dont forget the made for TV movie "The day after"
05:48.39drmessanoUm.. I wasnt describing a movie
05:48.44harry_vwith todays solid state world, EMP is still a serios threaght.
05:50.05drmessanoEMP is actually counterproductive
05:50.06harry_vyea, just detonate a thermo nuclear warhead in the ionosphere and rain a storm of supper charges elecotrons to earths surface. See how many day to day electronic devices survive and those that are hardned do survive.
05:50.10jayteeit's even more a threat today than it was in the 70's and 80's
05:50.11harry_vhow so
05:50.14drmessanoi doubt it will ever be a real threat
05:50.35drmessanoWell, shenanigans you plan to benefit from using EMP will be destroyed by EMP
05:51.03harry_vmy truck in general, will survive
05:51.24drmessanoYoure missing the big picture
05:51.42jayteeone 20 megaton warhead detonated 80 to 90 miles up over Kansas City could fry most of the electronics in the lower 48
05:52.10harry_vjaytee, was that from actuall DOD material?
05:52.13drmessanoWar is useless without the raping and pillaging
05:52.34echelonhow do services like ipkall and grandcentral get all those phone numbers?
05:52.44jayteeharry_v, of course not! I'd never divulge classified information
05:52.50harry_vheheh
05:52.55jayteeechelon, they steal them
05:52.57harry_vI know
05:52.59drmessanoThere will be nothing to pillage with EMP.. No electronic transfers, no real goods
05:53.13echelonseriously
05:53.17drmessanoJust damage... which will be effective, but have less of a point
05:53.24drmessanoechelon: They buy them
05:53.42echelondrmessano: just a one time fee?
05:53.48drmessanoNo
05:53.51harry_vthats why emergency supplies like food and fuel are good things to have on hand.
05:54.00echelondrmessano: so how can they provide them for free?
05:54.07drmessanoThey make money from them
05:54.12echelonhow?
05:54.16harry_vin fact, food would have been a good investment after the stock crash if you have the storage to hold it.
05:54.19drmessanoThey're not robin hood
05:54.23jayteeif we and Russia used only a third of our current nuke inventory on each other fighting a war there'd be only some bacteria surviving a year later
05:54.37drmessanofrom collecting intercarrier transfer fees for terminated calls
05:55.09coppicejaytee: rubbish. the human race is far from having the capability to cauterise the planet
05:55.15echelonoh
05:55.19drmessanoharry_v: Youre missing the whole point
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05:55.41harry_vi know alot more supplies would be needed.
05:55.42echelondrmessano: does the person receiving the calls get charged? or just their carrier?
05:55.47drmessanoharry_v: I am talking about MOTIVATION, not HOW GOOD YOUR TRUCK IS
05:56.26drmessanoThere's no MOTIVATION to EMP someone when you're not just disabling some capability of theirs, but destroying now the very thing you're after
05:56.48harry_vExplain motivation ?
05:56.51jayteeI'm watching this dumb sci-fi movie with Kristanna Lokken in it. I used to think she was so hot but then I found out she's a carpet muncher.
05:56.55drmessanoI just did
05:57.04*** part/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com)
05:57.08drmessanoOk, let me make this easy for you
05:57.09*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca)
05:57.49drmessanoIf we went to Iraq and said "We're gonna hit them where it hurts, and blow up all their Oil", how fucking stupid would that have been, seeing as how we came FOR THE OIL
05:58.02harry_vexactly
05:58.15drmessanoWe destroy the worldly assests of someone electronically, and we lose all their worldly assets we came to STEAL
05:58.21harry_vjust nuke the area with a nutron bomb.
05:58.23drmessanoSo EMP becomes STUPUD
05:58.26drmessanoSTUPID
05:58.30drmessano....
05:58.32drmessanoWTF dude
05:58.42harry_vkills everyone and does not destry the infrastructor
05:59.21drmessanoUm yes, it does
05:59.27jayteebetter to use a trojan worm that just transfers all their ones and zeroes to an undisclosed account in the Cayman Islands
05:59.47jayteeand then nuke em!
06:00.27harry_vI remember in my training I had to learn about the nuclear triad.
06:01.01[TK]D-Fenderjaytee: Which movie?
06:01.03harry_vthen chemical training.
06:01.14drmessanoYou apparently dont get the whole "Defeat them and steal their shit" mentality if you think EMPing them is a smart move.. Good luck stealing their electronic funds, their research technology locked up in that shit you just EMPed, etc
06:01.20drmessanoThats very, very stupid
06:01.36[TK]D-Fenderjaytee: She was in BloodRayne which was a total piece of garbage
06:02.11harry_vanything underground though, would be protected.
06:02.25drmessanoYeah.... about that
06:02.26harry_vor in a feraday cage
06:02.30drmessanoEverything overground would not be
06:03.16harry_vmost usaf and who what other millitary aircraft are hardened against a emp attack.
06:03.35harry_vAnywa on to another subject
06:03.52drmessanoYes, please
06:04.31drmessanoBefore we start talking about the best way to recover a fumble in a football game is by blowing up the actual football
06:04.40drmessanoduh
06:05.28harry_vprobebly more low level localized emp attack would be effective
06:08.55coppiceEMP's da bomb!
06:09.03drmessanolol
06:10.10drmessanoThat reminds of a line..
06:10.17drmessanoThat reminds me of a line..
06:10.27drmessanoSomething about the horse you rode in on..
06:10.30drmessanoOh wait, ha
06:10.46*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
06:11.05drmessanoSo whats why we HATE australia
06:11.11drmessanooh heh, hi Maliuta
06:11.17drmessanothats*
06:11.37drmessanotoo.... crosseyed...
06:12.03drmessanojaytee: I loaded 1.6.1 on another box, it failed miserably as well
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06:15.38[TK]D-Fenderdrmessano: 1.6.1 from beta, or 1.6.0.1?
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06:20.03jayteedrmessano, you could always try using Subversion and loading from the latest from trunk instead of one of the branches :-)
06:21.09jayteemaybe by some miracle they've got it fixed already in trunk
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06:21.49[TK]D-Fenderjaytee: So which movie was it?
06:22.16jaytee[TK]D-Fender, that's on now?
06:22.36jayteeBloodrayne
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06:22.49TrentCreekanyone use FreePBX with asterisk2billingg?
06:22.55jayteeshe's the one that played the new terminator in T3
06:23.52[TK]D-Fenderjaytee: Yeah, it was craptastic
06:24.05jayteeI watched the new director's cut version of Blade Runner earlier. I think Deckard was replicant himself and didn't know it.
06:24.14[TK]D-Fenderjaytee: She was smoking hot in Martal Kobat:The Series which is what launched her
06:24.37[TK]D-Fenderjaytee: the Deckard question is a very open-ended point.  Thats the best part
06:25.01drmessanoOh god, spoilers
06:25.09drmessanoHogwarts was really an STD
06:26.33jayteelol
06:26.56jayteeI love this one tshirt they had on tshirthell.com "Dumbledore dies on page 573"
06:27.37drmessanoMark Spencer accidentally invented asterisk when he compiled apache in the same directory as teamspeak
06:27.42jayteePhillip K. Dick was an awesome author
06:27.48jayteelol
06:29.17coppice"Slavery gets shit done" :-)
06:35.07jayteewell, I'm gonna try and get some sleep. got a flight to catch in the morning
06:35.12jayteelater all
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09:05.20Kdashey all
09:05.46Kdasi am getting "Apr 13 22:03:35 NOTICE[2553]: chan_sip.c:5473 sip_reg_timeout:    Â— Registration for '02490XXXX@nexcom.bg' timed out, trying again (Attempt #2)" i already have srvlookup=yes in sip.conf so whats wrong ?
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09:10.09Kdasanyone ?
09:12.44TrentCreekit means what it means
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09:17.06Kdasos everyone dead ?
09:17.31Kdass/os/is
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09:20.54echelonhi
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09:24.01Kdasumm
09:25.41echeloni'm thinking about purchasing this.. http://www.grandstream.com/ht286.html
09:26.17echelonafter i enter all the sip credentials, i can just dial any ptsn # as i would with a regular line?
09:30.44Kdasi am getting "Apr 13 22:03:35 NOTICE[2553]: chan_sip.c:5473 sip_reg_timeout:    Â— Registration for '02490XXXX@nexcom.bg' timed out, trying again (Attempt #2)" i already have srvlookup=yes in sip.conf so whats wrong ?
09:38.15TrentCreekJUST WHAT IT SAYS
09:38.44TrentCreekseems to be it's got a problem logging in
09:38.52TrentCreekor a problem with that domain name
09:38.59TrentCreekor could be a firewall problem
09:40.20TrentCreektry turning on SIP debug
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10:56.54TrentCreeknope
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11:25.53hi365any polycom gurus here?
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11:42.01feedscan asterisk convert sound from .wav? I mean automated messages like demo-echotest
11:53.17hi365yes, if its the correct wav
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13:43.14hi365wonders if he can set a dhcp server to do inform ONLY (i.e. no ip address)
13:44.04mankashwhat is sip show registry
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14:11.54ecmHello, all.
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15:38.49jayteemorning russell
15:39.01jayteemornin [TK]D-Fender
15:39.16[TK]D-Fender*yawn
15:39.58jayteedon't do that! I just everything on the breakfast buffet menu but I can't nap or I'll miss my flight.
15:40.01riddleboxmorning guys
15:40.17riddleboxjaytee, where you at?
15:41.03jayteeHuntsville
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15:44.45ThOr101Can someone just confirm the following statement:  In order to get zaptel kernel mods, you can't just install the zaptel packages (onto Fedora) you need to download the source, and compile.  Because the zaptel packages don't contain the kernel mods.
15:52.18jayteedon't know about Fedora way back I installed zaptel from packages but the kernel mods have to match the version of the kernel  you're running and most packages in repositories get "stale" fast.
15:54.14ThOr101fair enough.  Thanks.
15:55.34jayteecompiling isn't that big a deal, if you use Google I'm sure you can find a howto on the WIKI or some other site.
15:56.14ThOr101Oh, I've done it before, even with zaptel.  I resorted to compiling my own.  I just build a new system and was about to embark on the same path, and wanted to make sure I wasn't missing something obvious.  I'm compiling right now :-)
15:56.17jayteethere's one for CentOS on the WIKI which is very much like Fedora since they're both built based on Red Hat
15:57.26ThOr101I used to have it running, then tossed my old machine, and upgraded to 64 bit.  Things REALLY got wierd, so I yanked it.  Now I have some time, and an older machine that I can run 32bit on, so I'm giving it another go.
15:58.58hi365anyone good with linux dhcp?
15:59.10hi365im wondering: can I add option to be distributed to a class, or do they have to be in a pool?
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16:17.46nextimeis in * 1.4 STREAM FILE agi command stable, or i must use EXEC PLAYBACK instead?
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16:31.49tzafrir_laptopThOr101, that is a specific issue of Fedora. This is not the case with Debian and SUSE
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16:33.13ThOr101Thanks tzafrir_laptop .  Now I think I'm dealing with a bad card.  I seem to recall on my previous system that when I installed the kernel modules, the lights on the TDM422 card would pop on.  Not getting those lights, and just getting errors from the zt* commands.
16:33.27ThOr101I'm in a 5V slot with external power.
16:33.44ThOr101I removed all the addon cards.  Wierd.  I wonder if the card got friend somehow.
16:34.21tzafrir_laptopIIRC that module won't use a card without modules unless explicitly told so
16:34.40ThOr101Yeah, I put them in my zaptel.conf file.
16:34.45tzafrir_laptopwith: timingonly=1
16:34.48ThOr101That bit me last time I was doing this.
16:34.52ThOr101ohh, that's new.
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17:24.41rikstais murf here?
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17:41.41lesouvageHas anyone tried this phone http://www.globalsourcesdirect.com/servlet/the-1977/VoIP-Phone-For-SIP-fdsh-IAX2/Detail
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17:44.59root52lesouvage: No but the price looks right ;-)
17:46.00WimpManHmm. Seen a big bunch of them on ebay for 19.99.
17:46.20lesouvageroot52: with the sip and iax2 capabilities, all the promised features and the price it looks very interesting. And the phone looks just as ugly as all the other sip phones
17:47.24root52hahahaha interesting how i do not see a brand name.
17:48.18lesouvageroot52: I think the idea is to brand them yourself.
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17:48.59lesouvageroot52: like "take root52, the different way to go"
17:49.14[TK]D-FenderThats an ATCOM phone... PA1688 chipset flaming piece of shit :)
17:49.23ManxPowerThose look like those horrid little phones from China that use some specific chipset that someone (drunken college kids, I think) hacked up some untested IAX2 firmware for them.
17:49.47ManxPowerThat's it, the PA168 chipset.
17:49.58[TK]D-Fender1688 <- the 88 wasn't a typo :0
17:50.27ManxPower[TK]D-Fender: Odd, I thought it was PA168.
17:50.58ManxPowerI bought one one time.
17:51.06[TK]D-Fenderits a crap phone... plasitcy low feature pile of scarp
17:51.12[TK]D-Fenderscrap even
17:51.19ManxPowerOf all the phones I've owned that is the only one that I actually threw into the trash.
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17:55.18lesouvageManxPower and [TK]D-Fender: thanks for sharing your experiences. Ordering a full container doesn't sound like a good idea ;-)
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18:01.36[TK]D-Fenderjaytee: Quick flight...
18:10.27jayteeI'm in the smoking lounge at Huntsville waiting for my flight
18:11.16[TK]D-Fenderjaytee: Ah ah...  I just got in from picking up a Minolta 5000 w/ 50mm lens & flash for 50$.  The lens alone I'll likely resell for about 120$ :)
18:11.31[TK]D-Fenderjaytee: in the office now to continue on my ^%#$ing budget
18:11.36[TK]D-Fenderheh
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18:12.20Carlos_TicoDoes anybody knows if this a problem with pap2 configuration or asterisk?
18:12.20Carlos_TicoI have 1 remote pap2 that it registers ok, can make and receive calls but for no reason loses its registration randomly, sometimes minutes, or hours.
18:13.29jayteeCarlos_Tico: if the time interval for losing the reg varies widely it sounds like a flaky NIC in the pap2
18:13.56Carlos_Ticotried with two differents ATA
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18:14.01Carlos_Ticoone pap2 and one spa3102
18:14.20Carlos_Ticosometimes happens after the fist registration ends thats it
18:14.28jayteeCarlos_Tico: but you could try setting the registration interval to a low value so it refreshes more frequently
18:14.28Carlos_Ticowell most of the time
18:14.37Carlos_Ticonothing happen
18:14.40Carlos_Ticothe same
18:14.51Carlos_Ticoand in the ata debug says AUTH failed
18:15.29JymmmEMCDont mean to sound dumb, but If I was to setup asterisk, how do I get a PSTN number? Really just need a single DID if that makes a difference.
18:16.18Carlos_Tico[jaytee]
18:16.23Carlos_Ticoonly happens on the remote ata....
18:16.29Carlos_Ticothe network ata works perfect
18:16.50jayteeCarlos, if you're behind a NAT you need to set qualify=yes and canreinvite=no
18:17.51jayteeCarlos_Tico: that's assuming your pap2 is on the other side of a NAT
18:18.50hi365_mhow much of the polycom call features (cfw, dnd, etc) are suppported by asterisk? I couldnt even get my polycom blf's to distinguish between busy and ringing. is that the way it should be?
18:18.59Carlos_Ticolet me see
18:19.10Carlos_Ticoyes its a remote ata
18:19.46Carlos_Ticoyes i have those parameters set as qualify and no
18:21.00jayteeis the network link through an ISP or a private circuit?
18:21.21Carlos_Ticoits linked via ISP
18:23.23[TK]D-FenderJymmmEMC: How do you want to receive this DID?
18:24.09JymmmEMC[TK]D-Fender: TCP/IP (sorry, I dont exactly understand the question)
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18:24.57[TK]D-FenderJymmmEMC: Ok, from a VoIP provider.  Here :
18:24.59[TK]D-Fender~itsp
18:24.59jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
18:25.02[TK]D-Fender~itsplist-us
18:25.03jbot[~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, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
18:25.04[TK]D-Fender~itsplist-ca
18:25.04jbot[~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca
18:25.45Carlos_Tico[jaytee] and when i try to set up a host=myname.ath.cx ====> works well
18:25.57Carlos_Ticobut when that address changes ip asterisk cant resolve it
18:26.25JymmmEMC[TK]D-Fender: Ah, thank you. I have Skype right now, which would work fine for my needs, just not sure if there are SIP gateway voo doo magic to use it
18:26.49[TK]D-FenderJymmmEMC: To use Skype?  No, Skype is its own little world...
18:27.16JymmmEMC[TK]D-Fender: =) I just need a PSTN number to do some remote control is all.
18:27.32jayteeCarlos_Tico: if your wan link is getting hammered that could interfere with with keeping the pap2 registered. You might also check with your ISP about QoS to prioritize your traffic and make the necessary changes on your pap2.
18:27.43[TK]D-FenderJymmmEMC: Ok, but this leaves Skype completely out of the picture, just FYI
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18:28.14UnlockgodHey, wondering if someone could help me with a nat priblem
18:28.39JymmmEMC[TK]D-Fender:  i understand, google mentioned a couple of interfaces is all  (http://www.google.com/search?q=asterisk+skype)
18:28.40drmessanounlockgod?
18:28.49Unlockgodhi
18:29.01drmessanoWhat did you unlock?
18:29.04[TK]D-Fender~skype
18:29.05jbot[~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option.
18:29.07drmessanoand how was it godlike?
18:29.07[TK]D-Fender^^^
18:29.29Unlockgodmobile phones (cell phones)
18:29.31[TK]D-Fenderreboxes his Ark Of The Covenant
18:29.34Unlockgodjust a name used for years
18:29.38JymmmEMC[TK]D-Fender: cool. lol
18:29.54Unlockgod:p
18:29.57Carlos_Tico[jaytee] what do you mean wan getting hammered ?
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18:30.35JymmmEMC~SkypeForAsterisk
18:30.36jbot[~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details.
18:31.18[TK]D-FenderJymmmEMC: Its locked off, and "pay-per-channel"
18:31.25JymmmEMCew
18:31.26[TK]D-FenderJymmmEMC: Not available just yet
18:31.43[TK]D-FenderJymmmEMC: You can basically ditch Skype with SIP you know...
18:31.53jayteeCarlos_Tico: I mean if your bandwidth utilization for your ISP link is high due to other traffic that might interfere with registration
18:32.02[TK]D-FenderJymmmEMC: Skype is for little kiddies and the otherwise VoIP-ignorant
18:32.25JymmmEMC[TK]D-Fender: to me, it's $60/yr which is fine.
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18:32.42[TK]D-FenderJymmmEMC: $60 gets you what exactly?
18:33.40JymmmEMC[TK]D-Fender: a pstn number and unlimited inbound calling.
18:34.03[TK]D-FenderJymmmEMC: If you already have this, what do you want * for?
18:34.21tzafrir_laptopJymmmEMC, take a look at ipkall.com
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18:35.02Carlos_Tico[jaytee] is there a way that asterisk can resolve if a dyndyns change the ip .....
18:35.15black187Hello, anybody here with any experience with TLS setup on Asterisk?!?
18:35.38JymmmEMC[TK]D-Fender: I have a box in a datacenter that has a ham radio gateway. and I'd like to add a pstn number to perform some remote control.
18:35.50drmessanoblack187: you got TCP working?
18:36.04[TK]D-FenderJymmmEMC: Well that sounds like something to do... sure.
18:36.12black187nope, still stuck with SSL cert error on Asterisk :(
18:36.20*** part/#asterisk Unlockgod (n=bob@195.149.30.143)
18:36.23black187didn't even try the TCP connection
18:36.27Carlos_Ticoaytee ¦ Carlos_Tico: I mean if your bandwidth utilization for your ISP link is high due to other traffic that might interfere with registration ----> The unique device in the remote location is ATA
18:36.39Carlos_Ticoand a router of course.....
18:37.52JymmmEMCtzafrir_laptop: thank you!
18:38.55JymmmEMCDoes asterisk have a voice response /DTFM decoder feature?
18:39.01[TK]D-FenderJymmmEMC: Yup
18:39.16[TK]D-FenderJymmmEMC: IVR's are basic stuff
18:39.19black187Did anybody succesfully setup tls and tcp. I tried with different certificates, but none work (allways got error: SSL cert error - and the certificates I made were selfsigned and commonName was the same as IP of the computer with Asterisk...) - Is TLS and TCP on Asterisk tested?
18:39.26[TK]D-FenderJymmmEMC: Give the book a  good read
18:39.28[TK]D-Fender~book
18:39.28jbotmethinks book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
18:39.30[TK]D-Fender^^^
18:40.22JymmmEMC[TK]D-Fender: Ok, now the biggy.... Could I shove all this into a wrt54gl with OpenWRT and still get/have DTMF/voice response? (crosses fingers)
18:40.51[TK]D-FenderJymmmEMC: Sure
18:41.07JymmmEMChawt damn!  =)  I prefer hw over sw solutions =)
18:41.32[TK]D-FenderJymmmEMC: its still hardware... jsut cause its on flash and not a spinning HD doesn't change that fact :)
18:41.47[TK]D-FenderJymmmEMC: Now if you SOLDERED it.. THEN it would be a "hardware" solution ;)
18:41.54JymmmEMC[TK]D-Fender: No, no, I *LIKE* hardware solutions
18:42.03[TK]D-Fenderstill sofware*
18:42.18[TK]D-FenderJymmmEMC: Just nit-picking on you.. no worries..
18:42.24JymmmEMCOk, non-moving hardware... how's that =)
18:42.30[TK]D-FenderJymmmEMC: For what I'm sure your limited needs are it should do OK
18:42.41[TK]D-FenderJymmmEMC: Yeah that pretty much says it
18:43.05[TK]D-FenderJymmmEMC: faster-booting, non-spinning, cheap all-in-one box solution.
18:43.33JymmmEMCyou wouldn't know if the DTMF supports: 09*#ABCD by chance would ya?
18:44.49[TK]D-FenderJymmmEMC: It does
18:44.56[TK]D-FenderJymmmEMC: Full set
18:45.19JymmmEMCoh awesome!
18:45.32[TK]D-FenderJymmmEMC: not 100% sure on A-D for the hidden row as I've never seen it actually used, but I'm pretty sure its there
18:45.48[TK]D-FenderJymmmEMC: 4x3 is assured
18:46.44JymmmEMC[TK]D-Fender: Ok, I can research that, not really needed for inbound, but if I use asterisk from the radio side could add extra features.
18:48.07JymmmEMC(like autopatch, if you know what that is)
18:48.58[TK]D-FenderJymmmEMC: Digits are digits, more you can support the more meaningful & simpler overall entry can be.
18:50.35JymmmEMC[TK]D-Fender: Well, it's more of a (somewhat) security feature. Most phone don't have 4x4 but ham radios do =) Some silly war dialer kiddie playing around sorta thing.
18:51.50jayteeCarlos_Tico: sorry, I got into a conversation here at the airport lounge with a NASA engineer.
18:52.15Carlos_Ticoits ok
18:52.33Carlos_Tico[jaytee] is there a way that asterisk can resolve if a dyndyns change the ip .....
18:52.33JymmmEMC[TK]D-Fender: Thanks for the help/info, much appreciated!
18:52.43[TK]D-FenderJymmmEMC: np
18:53.23jayteeCarlos_Tico: not that I'm aware of as far as a built in function in *
18:53.59jayteei've got to logoff because the crappy battery in this Dell is about to die
18:54.08jayteelater all
18:54.14Carlos_Ticook thanks for all your help
18:54.52[TK]D-FenderCarlos_Tico: * behind NAT needing to know if IP changed?
18:55.19Carlos_Ticoyeah
18:55.38Carlos_Ticomy remote ata its with host=myname.ath.cx
18:55.53Carlos_Ticobut when it changes asterisk cannot resolve it ...
18:55.56[TK]D-FenderCarlos_Tico: You should setup your DynDNS client, and then do "externhost=my.dyndns.org" and "externrefresh=60" (seconds)
18:56.07[TK]D-FenderCarlos_Tico: You need the "externrefresh"
18:56.16[TK]D-FenderCarlos_Tico: that sets how often * rechecks it
18:56.26Carlos_Ticowhere to put that ... ?
18:56.38[TK]D-FenderCarlos_Tico: right beloh your externhost entry
18:56.41[TK]D-Fenderbelow*
18:56.52Carlos_Ticoin the trunk config
18:56.55[TK]D-Fender(under [general] of course)
18:56.56Carlos_Ticoright
18:56.58[TK]D-FenderCarlos_TRUNK?
18:57.16Carlos_Ticoor in the sip_nat.conf
18:57.20[TK]D-FenderCarlos_Tico: this is a [general] setting.  *'s situation is not trunk-dependant
18:57.32[TK]D-FenderCarlos_Tico:sip_nat.conf
18:57.36Carlos_Ticook
18:57.41[TK]D-FenderCarlos_Which of course tells me you're using FreePBX...
18:58.00Carlos_Tico:)
18:58.40[TK]D-FenderCarlos_Tico: But thats where it goes.  You should ahve "canreinvite=no", "nat=yes", "externhost=yourhosthere", "externrefresh=60"
19:01.13Carlos_Ticook
19:01.15Carlos_Ticolet me try
19:02.20*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
19:02.26Carlos_Ticobut that refreshes my host .... not the remote ATA host
19:03.11*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
19:04.19*** join/#asterisk jaytee (i=cea6ce22@gateway/web/ajax/mibbit.com/x-d8294305512ce823)
19:04.32[TK]D-FenderCarlos_Tico: Your ATA has its own dyndns?
19:04.37*** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net)
19:04.49Carlos_Ticoyes
19:04.52[TK]D-FenderCarlos_Tico: the ATA should be registering to YOU and you should be setting its register interval.
19:04.53jaytee\o/ found an A/C outlet in the smoking lounge to charge the battery
19:05.11[TK]D-FenderCarlos_Tico: You should never be specifying a host for them.  always "host=dynamic"
19:05.24[TK]D-FenderCarlos_Tico: Go into your ATA and chage the register interval.
19:05.33hi365_mhow much of the polycom serer-side call features (cfw, dnd, etc) are suppported by asterisk?
19:05.37Carlos_Ticoi did but i lost the registration with that
19:05.45[TK]D-FenderCarlos_Tico: How?
19:06.01*** join/#asterisk JackEStorm (n=no@ip24-252-118-155.no.no.cox.net)
19:06.55Carlos_Ticokeeps loosing the registration
19:07.12Carlos_Ticoso thats why in host i changed to dyndns
19:07.33[TK]D-FenderCarlos_Tico: that makes no sense.  a remote ATA should never have to care where ti is.
19:08.11[TK]D-FenderCarlos_Tico: how do you "lose" a registration?  * should not be a moving target, so all you should have to do is increase your register frequency.
19:08.46Carlos_Ticoin the syslog serv of the ata says Auth Failed .....
19:09.01drmessanoATA can be behind any nat
19:09.01[TK]D-FenderCarlos_Tico: that should have nothing to do with ANY IP.
19:09.07drmessanoany IP
19:09.15[TK]D-FenderCarlos_Tico: Auth failed = you set the auth up wrong.
19:09.15drmessanoThats why it REGISTERS
19:10.05Carlos_Ticobut everyting is fine... i just double checked....
19:10.17drmessanoAsterisk says its not
19:10.19drmessanoI believe it
19:10.20Carlos_Ticoi reset it registers then after a minut
19:10.31Carlos_Ticoits not register anymore ....
19:11.08drmessanoqualify is off for the extension
19:11.32Carlos_Ticoqualify is yes
19:11.48[TK]D-FenderCarlos_Tico: Your IP can't be changing every MINUTE.
19:12.05drmessanoFactory reset the ATA
19:12.07drmessanoThen reconfig
19:12.08[TK]D-FenderCarlos_Tico: You have really misinterpreted your problem.
19:12.08Carlos_Ticono the ip its not changing at all
19:12.12drmessanoWith 3 options
19:12.17drmessanousername, pass, proxy
19:12.18Carlos_Ticoi did....
19:12.20Carlos_Ticofactory reset
19:12.21Carlos_Ticopass
19:12.26[TK]D-FenderCarlos_Tico: then how can DyDNS have anything to do with fixing your probelm?
19:12.27Carlos_Ticoproxy ... and the same...
19:12.34[TK]D-FenderCarlos_Why were you lokoing at it in the first place?"
19:13.35Carlos_Ticojust to register a remote ata with *
19:14.00[TK]D-FenderCarlos_Tico: again, your ATA does not need this
19:14.35[TK]D-FenderCarlos_Tico: use imagebin to paste a screenshot of your ATA config and a pastebin of *'s SIP DEBUG for your failed login attempts./
19:14.36*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
19:15.19Carlos_Ticook wait
19:18.38*** join/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net)
19:18.50Carlos_Ticohere you are
19:18.51AndyMLmorning drmessano, [TK]D-Fender
19:18.56AndyMLmorning Carlos_PHX
19:18.57AndyMLerr
19:19.01AndyMLCarlos_Tico:
19:19.01Carlos_Ticohi :)
19:19.11Carlos_Tico[[TK]D-Fender]
19:19.21AndyMLguys - I've been working with Carlos so he's asked me to come over here and help him explain his problem.
19:19.22Carlos_Tico[AndyML] can explain better the situation....
19:20.05*** join/#asterisk zydoon (n=zydoon@41.225.159.36)
19:20.14drmessanomorning
19:20.27drmessanoWell
19:20.29*** part/#asterisk zydoon (n=zydoon@41.225.159.36)
19:20.31drmessanoIf I had to take a guess
19:20.36AndyMLThis Linksys ATA is behaving strangely re: registration. the first registration attempt succeeds and the device works for as long as the timeout is set for. After it expires though, further attemps fail.
19:20.40drmessanoThe far end firewall is a POS for SIP
19:20.50*** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl)
19:20.52AndyMLfar end being where the ATA is?
19:20.55drmessanoyes
19:21.01AndyMLit IS the ata...
19:21.05drmessano...
19:21.07AndyMLwhich doesn't make you wrong
19:21.14drmessanohe said hes using a PAP2?
19:21.14AndyMLbut its ironic at least
19:22.20AndyMLCarlos_Tico: is that ATA the router for that network?
19:22.20Carlos_Ticono
19:22.20AndyMLInteresting.
19:22.20Carlos_Ticoonly the ata part
19:22.20drmessanoSo what is the router?
19:22.20[TK]D-Fender...
19:22.27AndyMLbefore we get too deep - just one more piece of explaination
19:22.27Carlos_Ticothe router is a WRT300N Linksys with DDWRT sp1
19:22.28[TK]D-FenderSwow us the FAILURE.  Too much talk, not enough show.
19:22.48drmessanoAre you on Cable or DSL?
19:22.52AndyMLCarlos_Tico: dig up the screenshot of that auth failed from the ATA's logserver
19:22.57AndyMLDSL
19:23.04Carlos_TicoDSL is the remote ATA .....
19:23.09[TK]D-Fender?!?!
19:23.21drmessanoIs the router doing the PPPoE?
19:23.28drmessanoThe WRT300N?
19:23.41*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
19:24.05Carlos_Ticoyes
19:24.08*** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com)
19:24.11Carlos_Ticothe router is doing PPPoE
19:24.12drmessanook
19:24.12Carlos_Ticohttp://picpaste.com/Andy.jpg
19:24.29AndyML(After the first registration expires, asterisk denies further registration attempts. - I'll login and setup a test scenerio to get the sip debug dump for it...)
19:25.08drmessanoWill other devices work?
19:25.13drmessanoOther locations?
19:25.56Carlos_Ticoyes the x-lite works perfect
19:26.13drmessanobehind the same firewall?
19:26.17Carlos_Ticoyes
19:26.18drmessanosame extension?
19:26.38Carlos_Ticobehind the same firewall other extension ata is 4000 x-lite is 3000
19:26.46[TK]D-FenderAndyML: Lets make this clear.  The approach to solving this has been beating around the bush with a weed-whacker.  You need to buckle down on it.  enable SIP debug and show the succeeding entry, and then the failing entry.  PB the devices sip.conf tnry and pics of the ATA's config
19:27.08drmessanoCarlos_Tico: Blow up the 4000 entry and recreate it
19:27.35[TK]D-FenderAnd yes, take out the 4000 out of the picture and see if things start working.
19:27.48drmessanoUse the ATA on 3000
19:27.53Carlos_Ticook
19:28.12jayteestill ought to pastebin a sip debug of the fail
19:28.40drmessanoHes probably got something set wrong in the extension config in FreePBX
19:28.57drmessanoSo you blow it up, recreate it, dont fuck with it, and then bam it works
19:29.18*** join/#asterisk aksyn (n=aksyn@gw.na.nu)
19:29.25jayteeFreePBX? hmmmm
19:29.32drmessanoyes
19:30.04*** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-124-230.dsl.sil.at)
19:30.49AndyML[TK]D-Fender: will do.
19:31.21drmessanoI went to Goodwill yesterday on the "money" side of town..
19:31.23jayteedrmessano: this is interesting but I've gotta get to the gate for boarding time. Let me know later how it turns out, ok?
19:31.31drmessanoSince they have better second hand shit
19:31.32drmessanoOk, will do
19:31.38jayteethanks
19:32.00drmessanoThis woman was ARGUING over a pair of stained jeans she wanted to buy for $3 and cut up for something
19:32.09jayteedrmessano, [TK]D-Fender later guys
19:32.22[TK]D-Fenderjaytee: lter
19:32.24jayteeCarlos_Tico: good luck
19:32.25drmessanoThey offered her 10% off due to the stain, which is all goodwill will back down from stuff.. it's goodwill, afterall
19:32.41drmessanoSo she argued, and argued
19:32.44Carlos_Ticothanks jaytee
19:32.48Carlos_Ticojaytee
19:32.49drmessanoThen paid for it
19:32.52[TK]D-Fenderdrmessano: Goodwill neads a big stick to beat ill-will people with
19:33.01drmessanoThen went outside
19:33.08drmessanoGot into her $35000 SUV
19:33.12drmessanoand stormed out of the parking lot
19:34.25[TK]D-Fenderdrmessano: sombitch
19:34.44[TK]D-Fenderdrmessano: probably burned as much in gas slamming the pedal in anger leaving.
19:39.07drmessanoYes
19:40.03drmessanoHonestly, if something clearly overpriced I dont mind telling them
19:40.12drmessanoBetter to knock a buck off than to not sell it
19:40.18drmessanoBut They gave her the 10%
19:41.25*** join/#asterisk Daejeo (n=chatzill@118.219.208.186)
19:43.35Daejeowill it be useful for initiating the call (DISA)l via email/SMS on asterisk?
19:44.21[TK]D-FenderDaejeo: You tell us
19:44.50[TK]D-FenderDaejeo: I wouldn't consider it useful if I didn't have access to SMS / e-mail
19:45.07Daejeoit is very useful
19:45.13drmessanoor if my mail server didnt have thumbs
19:45.25Daejeowould you like to know how?
19:45.35Daejeoit is not useful americans
19:45.39Daejeofor*
19:46.23drmessanoRunning that through my IRC drama filter, I think he just called me useless
19:46.25Daejeoit is hard to find a DID numbers in south asia.
19:46.31Carlos_Ticook here is the ATA configuration .... http://picpaste.com/ATA.jpg
19:46.33drmessanospits in a bucker
19:46.41drmessanoCarlos_Tico: Did you put it on 3000?
19:46.46Carlos_Ticoyes
19:46.48Carlos_Ticonow is 3000
19:46.52drmessanoDid it work?
19:47.19Daejeobut most of the people have a cellphone phone
19:47.45drmessanoUg
19:47.51drmessanoUse auth ID NO
19:47.52Daejeoso sms can be used instead of DID number
19:47.58drmessanoGet rid of the auth stuff
19:48.10Daejeo[TK]D-Fender: any comments?
19:48.16drmessanoUse auth ID NO and get rid of the auth username
19:48.17drmessanoand it will work
19:48.21drmessanoYoure not using AUTH
19:48.54drmessanoIt registers, then fails to auth
19:49.08*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
19:49.15drmessanoProblem solved
19:49.58AndyMLfor my sake, can you explain that in a little more detail? I'm the one that set that auth stuff up because it was doing the same thing before...
19:50.15Carlos_Ticowe used yesterday with Andy
19:50.31Carlos_Ticoand before we were not using ....
19:50.49Carlos_Ticoso it was only for testing purpouses
19:50.58drmessanoAuthentication is autheticating.. not just registering
19:51.08drmessanoAuthentication is more a security mechanism
19:51.55AndyMLhow do you require it or not require it in asterisk?
19:52.02AndyMLauth=md5 ?
19:52.08drmessanoyes
19:52.13Carlos_Ticoi already tried with that no luck
19:52.14drmessanoauthusername I believe it is, etc
19:52.16Carlos_Ticosame happens
19:52.28drmessanoCarlos_Tico: Get rid of aith
19:52.30drmessanoauth
19:52.33Carlos_Ticook
19:52.34drmessanoDont try to make it work, dump it
19:52.52drmessanoYou need THREEE
19:52.53drmessano3
19:52.56drmessanoTRES
19:53.03drmessano3 params in the ATA, as I have told you for days
19:53.11drmessanoUsername, password, proxy
19:53.30AndyMLthats it
19:53.33AndyMLjust those three
19:53.34drmessanoYes
19:53.34Carlos_Ticoit was like that ... we changed only for testing purposes
19:53.47drmessanoanything else is tweaking
19:53.54drmessanoSome meaningful, some not
19:53.57drmessanoCodec, etc
19:54.08drmessanoBut 3 parms will get an ATA regged
19:54.22drmessano0 vs 1, work vs not work
19:54.50AndyMLdrmessano: for theory's sake, etc... this is a high latency situation. his ATA is between 300 and 2000ms and 35+ hops away. can you think of any tweaks that might be necessary to keep it registered in that situation?
19:55.14drmessanogood god
19:55.42drmessanoif theres that much latency, registration lag is a non-issue.. the device wont be USABLE
19:57.05AndyMLits usually at the lower end, and it works okay for their purposes (keeping family in touch, etc) we just need to keep it registering...
19:58.20root52If I register exten 2000 on server(externalIPaddress) I should be able to call 2000@serverExternalIPAddress right?
19:58.46*** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com)
19:59.11drmessanoRegister expires = 60?
19:59.17drmessanoWho set that?
20:00.12Carlos_Ticoit was just for testing
20:00.22Carlos_Ticothe original value was 600
20:00.45drmessanowhen I say things like "Factory reset it and put this in"
20:00.50drmessanoI am not kidding
20:01.59drmessanoWhat else is set for "testing"?
20:02.06Carlos_Ticonothing else
20:02.16drmessanoHow do I know that isn't another lie?
20:02.27drmessanoYou seem to like to not do things or say you've done things..
20:02.42drmessanoIt's impossible to work off known constants
20:03.13drmessanoI know these devices pretty well, as do others.. and by not setting things to known states, they cannot be troubleshot effectively
20:03.19drmessanoI guess you dont understand that
20:03.26Carlos_Ticosorry ... those were move only for testing
20:05.35AndyMLdrmessano: I haven't worked with this particular device much and made some changes...
20:05.44AndyMLregister expires = 60 was me to get your sip debug
20:05.51AndyMLerr [TK]D-Fender's sip debug
20:05.55AndyMLhttp://www.pastebin.ca/1258023
20:06.44AndyMLCarlos_Tico: are we able to reset this thing to defaults from the US? or would we need someone to do it there?
20:06.53[TK]D-FenderAndyML: wheres ther egister attempt?
20:06.54drmessanoI XML config all my linksys boxes now.. and I can tell you off the top of my head.. I set User/pass/proxy, timezone rule, and force the codec..
20:07.15*** join/#asterisk linagee (n=jalton@about/linux/staff/linagee)
20:07.18Carlos_Ticoi can call them to reset to factory defaults
20:07.25*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:07.34linageewtf? i was using polycom 2.2.0 and it took like 5+ minutes to load sip.ld!
20:07.46linageenow i upgraded to 3.1.1 and it takes under 30 seconds. huh??
20:07.47AndyML[TK]D-Fender: I hit 'sip set debug 1-pstn' and let it go until the registration expired. what do i need to do to get it to show the registrations?
20:08.05[TK]D-FenderAndyML: 1-psten?  this was 400 a few minutes ago?  WTF?
20:08.10[TK]D-Fender4000*
20:08.19drmessanoThats the PSTN FXO
20:08.21AndyMLits is an FXO/FXS device. neither side is workign
20:08.22drmessanoTheres an FXS
20:08.30drmessanoForget the FXO
20:08.39[TK]D-FenderOk, I've hit my limit of dealing with bait&switch debugging.
20:08.40AndyMLthey're both failing the same way at essentially the same time
20:09.28[TK]D-Fendergoes off to do something productive.
20:09.39drmessanoReset the boxes, set the extensions in FreePBX up again
20:09.51drmessanoPut in proxy, user, pass
20:09.53drmessanoand then test
20:09.53*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:09.58AndyMLwill do.
20:10.05*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
20:10.17drmessanoThere's too much tweaking going on here
20:10.21drmessanoNeed constants
20:10.39AndyML[TK]D-Fender: sorry for the headache I guess...
20:10.41*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
20:13.48drmessanoAndyML: If you called your doctor and told him you had a redish, blackish, white-ish, pinkish bump, he would tell you to call a doctor
20:15.48*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:17.26AndyMLI'm just saying - the sip debug would should the same errors for 1-pstn and for 3000 (since TK told him to switch it to 3000)...
20:17.42*** join/#asterisk Kdas (n=Kdas@c-98-207-95-143.hsd1.ca.comcast.net)
20:17.46*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
20:18.02AndyMLand it didn't show the registrations properly, so i asked how to enable that level of debug, but got no answer...
20:18.28Kdasi am able to make outgoing calls fine but i can't hear anything when i get a incoming call any ideas?
20:18.51[TK]D-FenderKdas: NAT involved somewhere along the way?
20:19.04[TK]D-Fender~sipnat
20:19.05jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
20:19.06[TK]D-Fender^^^^^
20:19.33Kdas[TK]D-Fender, yes its behind nat
20:19.40[TK]D-FenderKdas: read up
20:19.56Kdas[TK]D-Fender, ok thanks
20:23.21*** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr)
20:27.24*** part/#asterisk mgdm (n=michael@river.mgdm.net)
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21:03.57*** join/#asterisk monstertruck (n=monstert@174.149.193.81)
21:05.44*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
21:21.49*** join/#asterisk kimo_sabe (n=nick@zappa.azrackspace.net)
21:25.59root52Hi, When I make a call the softphone says "no route to destination" That is ok I know why that is happing. My question is should the not be at lease one line of output on the *CLI? Verbosity is set to 8. I ask because I do remember seeing this output in the past.
21:26.25kimo_sabehmm, g729 outgoign calls are dropping on me when the person picked up. What am I missing?
21:27.06kimo_sabeI bought a license, and it works for inbound calls
21:27.39[TK]D-Fenderroot52: Depends if * is accepting the call at all or not.  Enable SIP debug and see
21:28.42root52Ahhh good call
21:30.19kimo_sabethe asterisk console isn't giving me any hints as to why it's dropping the call
21:30.52kimo_sabe* is SIP Declining the call
21:31.18*** join/#asterisk arpu (n=arpu@chello084114022060.14.vie.surfer.at)
21:32.14[TK]D-Fenderkimo_sabe: Go looka t your call at verbose 10, sip debug enabled.
21:32.31kimo_sabe[TK]D-Fender: ah
21:34.10[TK]D-FenderIt was a sad day when the Lone Ranger realized that "kimosabe" translated as "horse's ass" :p
21:35.52kimo_sabe[TK]D-Fender: indeed :)
21:37.33kimo_sabehmm, still just "Zap/1 answer SIP/blah"
21:37.40kimo_sabehangup Zap/1
21:39.46[TK]D-Fenderkimo_sabe: Still you not showing us the call in a pastebin with the kind of debug you were suggested to do.
21:40.16kimo_sabe[TK]D-Fender: oh, you wanted it in pastebin, one sec
21:40.30[TK]D-Fenderkimo_sabe: Are you expecting help when you show nothing?
21:41.03kimo_sabe[TK]D-Fender: I was hoping for some hints to expose the right debug information so I can figure it out
21:41.22[TK]D-Fenderkimo_sabe: I did.  Told you verbsoe 10 & sip debug.
21:44.43*** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com)
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21:54.02kimo_sabehttp://pastebin.ca/1258225
21:57.24kimo_sabeum kay, so the console isn't showing everything that make it into the log
21:57.41*** join/#asterisk shinao1 (n=shinao1@62.173.48.42)
22:06.51[TK]D-Fender[Nov 16 14:51:38] WARNING[12687] channel.c: No path to translate from SIP/307-b856ca90(256) to Zap/1-1(68)
22:07.15kimo_sabe[TK]D-Fender: yeah, I saw that in the logs (but not the console I had been looking at).
22:07.17[TK]D-Fenderkimo_sabe: You say you bought a license.  Now show us that it is installed and available
22:07.33kimo_sabeppbx*CLI> show g729
22:07.33kimo_sabe0/0 encoders/decoders of 1 licensed channels are currently in use
22:09.08*** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au)
22:09.17kimo_sabebug 8781 is looking similar
22:09.21*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
22:11.52kimo_sabehmm, but that claims to have been fixed before 1.4.4 was tagged
22:20.31*** join/#asterisk monstertruck (n=monstert@174.149.193.81)
22:20.37[TK]D-Fenderkimo_sabe: Trya a call from 307 to something else that clearly requires tranascoding.
22:21.23baliktadCan someone have a look and tell me if this call is failing because of me or my provider?  http://pastebin.ca/1258285
22:24.50protocolsis there any good howto for nvfaxdetect?
22:24.50[TK]D-Fenderbaliktad: SIP/2.0 483 Too Many Hops <- badsetup on your end
22:25.03kimo_sabe[TK]D-Fender: hmm, neat. I can call the fax machine and listen to it squeal
22:25.27baliktadall I can see is the one hop from me to my provider.  Why are they claiming too many hops?
22:25.56[TK]D-Fenderbaliktad: PASTEBIN YOUR PEER ENTRY
22:25.59[TK]D-Fenderdarn caps
22:26.52baliktadhttp://pastebin.ca/1258296
22:28.44[TK]D-Fenderbaliktad: yOU'RE BEHIND nat, AREN'T YOU?
22:28.59baliktadI am
22:29.10[TK]D-Fenderbaliktad: You should have put "nat=no" for this entry then.
22:30.36baliktadsame result (yes, I reloaded sip module)
22:33.51[TK]D-Fenderbaliktad: Now PB your sip.conf masking only passwords
22:36.27*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
22:38.47kimo_sabewhat could be different between Zap/1 & Zap/28 that would stop g729 transcoding from working?
22:41.29baliktadok, once again, I'm an idiot
22:42.00baliktadI had a duplicate account in my sip.conf from an earlier copy/paste
22:42.01kimo_sabeand Zap/1 -> SIP/307 does work
22:43.12*** join/#asterisk jov4n (n=jovan@87.19.163.231)
22:43.14jov4nHi
22:49.35*** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca)
22:51.05drmessanoAnyone know is a 16vac power supply for a 480e is a "standard" of some sort of Aastra
22:51.10drmessanoif*
22:52.05drmessanoI have been trying to find one online.. thinking maybe all the 480s or maybe ALL their phones use the same supply
22:53.47drmessanoAh, think I found one
22:53.50drmessano$20 bleh
22:54.15drmessanoAnyone need an Aastra 480e? lol
23:06.09[TK]D-Fenderkimo_sabe: Any chance any other call could have been in progress using that license?
23:07.01*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
23:07.28*** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal)
23:13.01*** join/#asterisk CrazyTux (n=brandon@75.4.22.105)
23:18.10kimo_sabe[TK]D-Fender: this is the only phone with allow=g729, and there are no other active channels
23:18.55[TK]D-Fenderkimo_sabe: I suggst you do a channel dump prior, and post and test again
23:19.16kimo_sabe[TK]D-Fender: channel dump?
23:19.30[TK]D-Fendershow channels concise
23:20.38kimo_sabenothing before or after, this while ringing: Zap/1-1!from-zaptel!8913738!1!Dialing!AppDial!(Outgoing Line)!8913738!!3!0!(None)
23:20.41kimo_sabeSIP/307-b8778d90!macro-dialout-trunk!s!25!Ring!Dial!ZAP/g0/5208913738|300|W!5208886740!!3!0!(None)
23:21.31[TK]D-Fenderkimo_sabe: ok, I'm at a bit of a loss right now...
23:22.51interfaithquestFender, i can see your hard core with asterisk ! great
23:23.20interfaithquestso i heard the audio on twit.tv of mark spencer.. cool
23:24.41interfaithquestanyway.. are you using dundi ? i am wondering how to random users can find each other , call from * to * using iax for example
23:25.17interfaithquestdundi seems more of a gateway sharing system, than anything else ?
23:27.05kimo_sabeok, well, nobody call the patent nazi's on me, but I'm using the opensourced G.729 but still sticking to the 1-call of usage
23:31.28*** join/#asterisk postel (n=jp@wikimedia/Postel)
23:32.41[TK]D-Fenderkimo_sabe: Go ask them why it doesn't transcode then :)
23:33.12kimo_sabe[TK]D-Fender: well, thanks for you help. I'll just stick to the honor system and only use the 1 seat I paid digum for, but with the unrestricted codec
23:33.37kimo_sabe[TK]D-Fender: the OSS one works fine, I guess I was somehow hitting the limit with the digium codec with only 1 call
23:33.42[TK]D-Fenderkimo_sabe: You saying that the OSS one works where your paid one doesn't?
23:33.47kimo_sabe[TK]D-Fender: yup
23:33.54[TK]D-Fenderkimo_sabe: You should call Digium about that..
23:35.04*** join/#asterisk joobie (n=joobie@mx01.anric.com.au)
23:35.33kimo_sabe[TK]D-Fender: I'm thinking there might be some sort of weird double-allocation thing going on, maybe with this older version of *, or something FreePBX is doing. This will work for now and I'll recheck the digium version whenever we do a big update on this system
23:36.18*** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-081-096.dsl.sil.at)
23:36.56[TK]D-Fenderkimo_sabe: You've got a solution at least and in my own peronal way I agree with how you did it.  You paid for the rights (more or less).  Your bypass is pretty legit to me, but I would make best efforts to correct it in the expected manner
23:40.18[TK]D-Fenderinterfaithquest: DUNDi ?  meh...
23:48.45*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
23:48.49interfaithquestsome registry of asterisk servers ? for * to * calling
23:50.01drmessanoDundi is a good idea for that
23:50.06interfaithquesti guess one could simply click on a webpage for a click to call service
23:50.16drmessanoIntercompany
23:50.22drmessanoor load balancing in a way
23:51.00interfaithquestanyway.. why call unless you find the party of interest
23:51.42interfaithquestsomeway to post your * for voip connections
23:52.11drmessanoYou can post your URI
23:53.08interfaithquestyes , others have tried sip proxies.. where they give YOU ..there ID.. ha ha..who is in the middle.. of the universe ?
23:54.20drmessanoIm talking direct SIP dialing
23:54.23interfaithquestso if you have an * machine and have 4569 exposed or are exposed thru some iax proxy. .then presto.. the window to the world is open
23:54.25drmessanoTo your PBX
23:54.55interfaithquestsip is crippled by the NAT issues world wide
23:55.01drmessanoheh
23:55.05drmessanoSIP is just fine
23:55.20drmessano"NAT Issues" fall into the realm of improper setup
23:55.32interfaithquesttell that to the milliions of skype user
23:55.47drmessanoWhat does Skype have to do with SIP?
23:55.50drmessanoor NAT?
23:56.10interfaithquestif SIP did not have a scaleable barrier the world would use sip.. why is the internet  not today full duplex peer to peer for voice
23:56.22interfaithquestinstead the net is basically a web client universe
23:56.34*** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk)
23:56.39drmessanoSkype users dont have SIP connectivity due to the closed nature of Skype
23:56.47drmessanoHas nothing to do with SIP
23:57.14interfaithquestso where is the global universal protocol, the whole world can use for FOSSL voip ?
23:57.49drmessanoDunno, are you gonna invent it?
23:57.55drmessanoLemme know how it goes
23:58.00interfaithquestwill do
23:58.10interfaithquestit's time has come for sure
23:58.39drmessanoIm not sure how to break this to you
23:58.49drmessanoBut SIP URI dialing is well on its way to becoming a standard
23:59.05interfaithquestwell google seems to be happy pushing XMPP jingle for its part
23:59.14interfaithquestyes.. sip may break thru with ICE
23:59.20TrentCreekOkay..I got the trunk working. I used the example context provided by les.net
23:59.30drmessanoEver heard of a little company called Microsoft who has been pushing SIP dialing?
23:59.51interfaithquesttrue billy goat does have desire to rule the world

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