00:00.29 | mchou | lesouvage: please stop "helping" me |
00:00.54 | mchou | lesouvage: it's clear you have no idea what you're talking about |
00:01.08 | mchou | lesouvage: thanks but no thanks |
00:02.28 | lesouvage | Is there any second opnion, I really don't think I'm talking nonsense. |
00:02.48 | mchou | lesouvage: You are jabbering NONSENSE |
00:03.12 | interfaithquest | anyone tried chan_gtalk ? |
00:04.13 | mchou | interfaithquest: I have it active but I've never tried it. Wanna test? |
00:04.23 | interfaithquest | ok |
00:04.26 | lesouvage | mchou: I could pastebin you a working example of what you are trying to achieve. But it seems that you are not interested. |
00:05.38 | mchou | lesouvage: 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) |
00:10.30 | lesouvage | mchou: 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: ${CallerID(num)}) (cahnge s if needed) |
00:11.31 | *** join/#asterisk synchris (n=synchris@athedsl-156469.home.otenet.gr) |
00:11.34 | lesouvage | mchou: have a good live |
00:11.54 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
00:11.54 | *** mode/#asterisk [+o russellb] by ChanServ |
00:12.15 | lesouvage | . |
00:12.26 | raasdnil | [TK]D-Fender: hey, phones are all woking good. Got an issue with the fax machines. Problem is the user dialed 1414001181339153100 on the fax and asterisk is trying to dial: 1343194105031100,1. See http://www.pastebin.ca/1257290 for CLI output and extensions.conf |
00:12.46 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
00:13.13 | raasdnil | sorry, http://www.pastebin.ca/1257291 |
00:14.06 | [TK]D-Fender | raasdnil: -- Executing [1343194105031100@out-nec:1] Dial("DAHDI/31-1", "DAHDI/g2/1343194105031100,,Tr") in new stack |
00:14.13 | [TK]D-Fender | raasdnil: -- Called g2/1343194105031100 |
00:14.19 | [TK]D-Fender | raasdnil: there is no ",1" in there |
00:14.39 | raasdnil | yeah, but on the fax, I personally dialed 1414001181339153100 |
00:14.55 | raasdnil | somehow getting mangled.... :/ |
00:15.41 | raasdnil | oh... try the second pastie... i posted the wrong bit of extensions.conf (been up for a few hours now) |
00:15.41 | [TK]D-Fender | raasdnil: That would be so mangled that I wouldn't trust your telling me that it happend at first glance |
00:16.01 | raasdnil | [TK]D-Fender: I didn't believe the person either. |
00:16.02 | [TK]D-Fender | raasdnil: new PB please... make it the right one this time |
00:16.19 | raasdnil | http://www.pastebin.ca/1257291 |
00:16.40 | raasdnil | [out-nec] |
00:16.40 | raasdnil | exten => _X.,1,Dial(DAHDI/g2/${EXTEN},,Tr) |
00:16.52 | raasdnil | on the end |
00:17.09 | [TK]D-Fender | raasdnil: again I see no ",1" on the end of your dial |
00:17.30 | raasdnil | one sec |
00:17.31 | raasdnil | sorry |
00:18.16 | raasdnil | http://www.pastebin.ca/1257296 |
00:18.20 | raasdnil | that one is correct |
00:20.33 | Dr-Linux|home | [TK]D-Fender: tried alot but providing dial tone in AGI didn't work |
00:20.57 | [TK]D-Fender | Dr-Linux|home: ok/fine/sure |
00:21.11 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
00:21.41 | [TK]D-Fender | raasdnil: Look in that PB. still no ",1" as part of the dial. |
00:21.58 | [TK]D-Fender | raasdnil: Put. down. The. Crack. Pipe. (c) JerJer |
00:22.15 | raasdnil | [TK]D-Fender: but it's the only thing keeping me awake! :) |
00:22.31 | [TK]D-Fender | raasdnil: I would never do a drug named after a part of my ass <- |
00:22.35 | raasdnil | isn't the exten => _X.,1, the ,1 you are talking about :) |
00:23.05 | raasdnil | has some basic misunderstood here |
00:23.09 | [TK]D-Fender | raasdnil: "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-Fender | chars* |
00:23.43 | [TK]D-Fender | raasdnil: Paste the SINGLE line where you see * doing DIAL with those added chars |
00:25.11 | raasdnil | [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-Fender | NEXT!@@!@!!@ (c) BKW |
00:26.26 | interfaithquest | join #asterisk-dev |
00:26.32 | interfaithquest | oops |
00:27.08 | raasdnil | throws [TK]D-Fender another newbie |
00:27.26 | [TK]D-Fender | goes to hide the bodies |
00:28.08 | raasdnil | newbie: [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-Fender | reaches for his katana.... |
00:30.30 | raasdnil | [TK]D-Fender: ok I think it might be the fax machine. |
00:30.36 | raasdnil | the ,1 was a red herring |
00:30.58 | raasdnil | I dial 96927300 on the fax, the NEC prepends 1414 and into the * box comes: |
00:31.10 | raasdnil | <PROTECTED> |
00:31.32 | [TK]D-Fender | raasdnil: Seems to truncate 2 digits |
00:32.00 | raasdnil | if 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 |
00:32.07 | raasdnil | yeah... |
00:32.08 | raasdnil | weird |
00:32.19 | raasdnil | do you have to treat fax machines differently? |
00:32.37 | [TK]D-Fender | raasdnil: this is your PBX being retarded... "not our problem" |
00:32.39 | raasdnil | maybe I'll go put a POTS phone on that fax line and dial and see if the NEC system is playing funny buggers |
00:32.43 | raasdnil | heh |
00:33.03 | raasdnil | newbie: my pabx company said that the asterisk guys would know how to integrate though... :D |
00:33.26 | [TK]D-Fender | raasdnil: What else did your Rice Crispies say to you? :) |
00:33.43 | raasdnil | nyah... read it off the back of a pack of wheetbix :) |
00:33.48 | raasdnil | bbs |
00:35.19 | harry_v | 90s in cali = perfect for fires. |
00:35.50 | harry_v | I 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) |
00:37.32 | drmessano | Those services are mostly unreliable |
00:39.15 | harry_v | when 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) |
00:49.00 | drmessano | No |
00:49.07 | *** part/#asterisk seaq (n=seaq@98.227.60.190.host.ifxnetworks.com) |
00:52.40 | [TK]D-Fender | checkout time, later all |
00:56.15 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
00:59.27 | drmessano | So ummm |
00:59.39 | drmessano | About that Exchange UM + Asterisk |
00:59.51 | drmessano | Any way to see if TCP is working in Asterisk? |
01:00.02 | drmessano | other than knowing tcpenable=yes? |
01:00.09 | jaytee | got two tcp capable phones? |
01:00.32 | drmessano | What the hell is that gonna pro... oh, hang on.. |
01:00.45 | jaytee | set them up in * with canreinvite=no |
01:01.05 | drmessano | I got an ATA.. doing that now |
01:01.12 | jaytee | then 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) |
01:02.06 | drmessano | lol |
01:02.20 | drmessano | Im gonna do the two ports independent |
01:03.07 | jaytee | you mean one phone tcp and the other udp to make * do the transform? |
01:04.11 | drmessano | Um.. 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) |
01:09.18 | drmessano | Damnit 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) |
01:33.32 | drmessano | I cant register Windows Messenger to Asterisk with TCP |
01:33.39 | drmessano | It works with UDP |
01:33.58 | drmessano | Unless there's some other difference, TCP appears to be NOT working |
01:35.05 | drmessano | Oh |
01:37.15 | drmessano | Scratch that.. working now |
01:37.15 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.5) |
01:37.16 | drmessano | lol |
01:44.04 | *** part/#asterisk korihor (n=korihor@200-71-160-1.genericrev.telcel.net.ve) |
01:44.35 | drmessano | Jaytee |
01:45.31 | drmessano | Anyone using TCP with 1.6? |
01:46.00 | etfonhomey | How do you have Windows Messenger registering to Asterisk? |
01:46.01 | mvanbaak | nope ;) |
01:46.40 | drmessano | I set the SIP options and click connect? |
01:47.24 | etfonhomey | Where in Windows Messenger do you have the ability to set SIP options. |
01:47.26 | etfonhomey | ? |
01:47.32 | drmessano | Apparently I cannot make calls from UDP peers to TCP peers and vice versa through asterisk, which makes no sense |
01:47.47 | drmessano | Tools > Options > Accounts |
01:48.20 | drmessano | and TCP to TCP isn't working.. almost like TCP is enabled but.. not so much |
01:48.26 | etfonhomey | What version of Windows Messenger are you using? |
01:48.38 | drmessano | 5.1 |
01:48.51 | drmessano | The latest |
01:48.57 | drmessano | or should I say, last |
01:48.59 | etfonhomey | Ah. I have 4.7 |
01:49.06 | mvanbaak | use something decent |
01:49.21 | etfonhomey | I do and it's not a Microsoft product. |
01:49.23 | drmessano | Decent? |
01:50.03 | mvanbaak | yeah, something without an evil EULA |
01:50.11 | drmessano | Ah here we go |
01:50.36 | mvanbaak | ;) |
01:50.47 | coppice | the evil Eula should be a character in some epic tale of ancient Egypt |
01:50.55 | drmessano | I 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) |
01:51.24 | coppice | drmessano: they come free with every meaningful question |
01:51.27 | mvanbaak | have you tried the tcp stuff with some other client ? |
01:51.49 | mvanbaak | or hardphone ? |
01:51.59 | drmessano | I have neither |
01:52.39 | drmessano | Im trying to get unified messaging working with Exchange.. and I wanted to make sure TCP was at least *working*.. I can register just fine |
01:53.18 | mvanbaak | what's the error on asterisk cli ? |
01:53.34 | drmessano | CHANUNAVAIL |
01:54.07 | mvanbaak | try a sip debug |
01:54.08 | drmessano | I can call my IVR just fine with Windows Messenger using TCP |
01:54.16 | drmessano | Just not another peer |
01:54.49 | mvanbaak | and exchange um cant use udp ? |
01:55.00 | drmessano | lol no |
01:55.14 | drmessano | Forgetting exchange UM |
01:55.22 | drmessano | TCP device can reg and call IVR |
01:55.35 | drmessano | But not another phone thats using UDP |
01:55.39 | *** part/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
01:55.46 | drmessano | Sounds like something internal |
01:56.15 | mvanbaak | and this is true for non-messenger clients using tcp as well ? |
01:56.36 | mvanbaak | try disabling reinvite |
01:56.39 | drmessano | I have not tested any non messenger clients |
01:56.46 | drmessano | canreinvite=no is set |
01:56.55 | coppice | drmessano: 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. |
01:57.11 | drmessano | Ok, but its working with asterisk one way |
01:57.15 | etfonhomey | Guess I need some kind of tcpenable in my sip.conf? |
01:57.22 | drmessano | I can call my IVR |
01:57.25 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
01:57.41 | mvanbaak | coppice: that's because asterisk does not support utf-8 |
01:57.51 | drmessano | I can call my wakeup call app |
01:58.11 | coppice | no. its because asterisk was complaining about any trailing stuff |
01:58.13 | drmessano | So that leg is working 100% |
01:58.23 | mvanbaak | drmessano: have you tried the other way around ? |
01:58.35 | mvanbaak | using an udp device to call your tcp device ? |
01:58.41 | drmessano | ..... |
01:58.50 | drmessano | We keep going over this.. let me lay it out again |
01:59.06 | drmessano | TCP device calls IVR and internal apps - CHECK |
01:59.16 | drmessano | TCP device calls UDP device - FAIL |
01:59.23 | drmessano | UDP device calls TCP device - FAIL |
01:59.26 | drmessano | TCP device calls TCP device - FAIL |
02:00.17 | mvanbaak | where TCP device == ms messenger |
02:00.29 | drmessano | Why should that make a difference? |
02:01.10 | mvanbaak | gheh |
02:01.34 | drmessano | Isn't asterisk a B2BUA? |
02:02.09 | mvanbaak | well, every client has their own special stuff |
02:02.20 | drmessano | Sure, but this client is working |
02:02.21 | mvanbaak | some are compatible, some arent |
02:02.36 | mvanbaak | that's why I want to know if this is true with other tcp clients as well |
02:02.53 | drmessano | I dunno, I can run out and buy a polycom real quick |
02:03.00 | mvanbaak | that way we know if it's a general tcp error or just incompatibility with ms messenger |
02:03.07 | drmessano | ... |
02:04.20 | mvanbaak | anywayz, I'm going to bed |
02:04.25 | mvanbaak | Sun Nov 16 03:04:02 CET 2008 |
02:04.28 | mvanbaak | latero all |
02:05.38 | drmessano | Well, MS hate strikes again |
02:05.58 | drmessano | Obviously if it's an MS product involves, it MUST be at fault.. |
02:06.05 | drmessano | involved |
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) |
02:10.30 | Daejeo | i want to trigger the call by email |
02:10.55 | Daejeo | any doc to get some idea? |
02:11.12 | coppice | drmessano: who has said MS must be at fault, apart from you? |
02:12.51 | drmessano | mvanbaak actually made a couple comments implying such.. but no worries, trying eyebeam now |
02:15.30 | drmessano | Same problem |
02:15.48 | coppice | he suggested incompatibility. I don't think he actually said who's fault it might be :-) |
02:16.05 | drmessano | eyebeam does the same using TCP |
02:16.11 | drmessano | soooo |
02:16.29 | raasdnil | therefore... microsoft must be at fault :) |
02:16.38 | drmessano | of course.. |
02:16.40 | raasdnil | perfectly logical when you think about it :) |
02:17.05 | raasdnil | has flash backs of monty python witch scenes |
02:17.11 | drmessano | So apparently TCP isn't completely working in 1.6 |
02:19.52 | coppice | various people claim this to be true |
02:20.08 | drmessano | lovely |
02:20.55 | drmessano | I guess there is a bright side |
02:21.37 | jaytee | ms messenger isn't SIP |
02:21.42 | coppice | now you're delusional |
02:21.58 | drmessano | sure it is |
02:22.23 | drmessano | Windows messenger is SIP and is actually a pretty decent softphone |
02:22.37 | drmessano | Theres MUCH better |
02:22.46 | drmessano | But for a freebie |
02:22.51 | drmessano | and for testing |
02:23.02 | drmessano | not too bad |
02:25.13 | coppice | if its considerably worse than the very best softphone it must be awful :-\ |
02:25.36 | drmessano | HA |
02:25.59 | drmessano | Theres some halfway decent ones |
02:26.06 | drmessano | s/ones/one |
02:26.25 | jaytee | Office Communicator supports SIP but the early versions of MS Messenger used it's own protocol. When did they add sip support to it? |
02:26.32 | drmessano | 5.0 |
02:26.44 | drmessano | 5.1 |
02:26.54 | drmessano | LCS 2003 didn't have Office Communicator |
02:27.07 | drmessano | It's native client was Windows Messenger 5.0 and later 5.1 |
02:28.51 | drmessano | Windows 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) |
02:35.17 | jaytee | well, 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) |
02:36.57 | drmessano | Yeah well |
02:37.07 | drmessano | you got the ASS part right |
02:37.12 | drmessano | , pain in my |
02:38.34 | drmessano | I simply cannot call a TCP device |
02:38.59 | drmessano | grrr |
02:39.18 | jaytee | transport = tcp in the client's section of sip.conf? |
02:39.31 | drmessano | yep yep |
02:39.51 | drmessano | Trying on two different boxes here |
02:39.59 | jaytee | futures = porkbellies in commodity.conf? |
02:40.02 | drmessano | Im about to get some alcohol |
02:41.22 | drmessano | At least I know now that its not exchange |
02:46.05 | jaytee | how'd ya figure that? |
02:46.14 | drmessano | Eyebeam doesn't work either |
02:46.19 | drmessano | Neither does Messenger |
02:46.22 | drmessano | Same symptoms |
02:46.57 | jaytee | you mean testing to and from each other, not to Exchange |
02:47.16 | drmessano | yes |
02:48.12 | jaytee | maybe that's why they didn't cover sip tcp in the Asterisk Advanced class even though we used 1.6 |
02:48.19 | drmessano | lol |
02:48.43 | drmessano | See, the thing is |
02:48.57 | drmessano | TCP client <> Asterisk works |
02:49.07 | drmessano | TCP client <> Asterisk <> UDP/TCP Client does not |
02:49.19 | drmessano | Theres something missing |
02:49.36 | jaytee | they'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." |
02:49.45 | drmessano | lol |
02:50.15 | drmessano | 5 hour lab.. "Configure a TCP device with asterisk" |
02:50.20 | drmessano | "And?" |
02:50.23 | drmessano | "No, thats it" |
02:50.24 | jaytee | so TCP to * to access something like Playback(boss-is-an-asshole) works? |
02:50.34 | drmessano | yep yep |
02:51.06 | drmessano | Its not umm.. trans--... transgendering |
02:51.24 | jaytee | transforming |
02:51.25 | Maliuta | jaytee: of course you hacked it so the call is actually Payback(boss-is-an-asshole) :) |
02:51.53 | drmessano | transportingporting |
02:52.10 | drmessano | yeah, transforming.. Like a weak autobot |
02:53.13 | *** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net) |
02:53.17 | jaytee | I'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 |
02:54.10 | Micc | I can't believe all the problems I've been having was from a stupid sql trace file being too big. |
02:55.16 | jaytee | and 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." |
02:56.50 | drmessano | http://bugs.digium.com/view.php?id=13117 <-- tada |
02:59.38 | jaytee | so there's a patch then |
03:01.03 | drmessano | A 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) |
03:05.44 | jaytee | drmessano, check this one out too, http://bugs.digium.com/view.php?id=13523 |
03:08.54 | dramman | I'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) |
03:09.43 | jks | what is the difference between state "Ring" and "Ringing" for newstate events on the manager interface? |
03:10.31 | dramman | I'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 |
03:10.39 | harry_v | jks, what versio are you running? |
03:10.53 | jks | harry_v, 1.4.21 |
03:10.57 | harry_v | good |
03:11.13 | jks | harry_v, ? |
03:11.15 | dramman | Or should the register string be ..../engin_in ? |
03:12.45 | dramman | If 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) |
03:17.16 | baliktad | your register string looks OK as is |
03:17.32 | baliktad | you need an incoming context in your dialplan for the call to be routed to |
03:17.40 | baliktad | then you can handle the call however you see fit |
03:19.11 | dramman | Can you see anything wierd/wrong in this? http://pastebin.com/d55c8876 |
03:21.17 | dramman | For incoming calls, it should be going to [DID_engin_36_in] |
03:21.17 | baliktad | so right now your incoming calls from engin get answered and then just ring extension 7000 |
03:22.24 | baliktad | not sure why line 209 has extension s with a period after it |
03:22.59 | dramman | no - "chan_sip.c:13885 handle_request_invite: Call from '' to extension '0290115436' rejected because extension not found." |
03:23.21 | baliktad | paste a sip debug |
03:25.11 | dramman | http://pastebin.com/d1b43d037 |
03:26.00 | dramman | I've actually commented out line 209 |
03:26.19 | baliktad | line 43 - your incoming call isn't getting matched to a sip peer |
03:26.25 | baliktad | so the call is going to your default context |
03:26.36 | baliktad | which of course doesn't have an extension matching the incoming number |
03:27.10 | dramman | shouldn't it match [engin_36_in] at line 49? |
03:27.53 | dramman | (line 49 of the first (config) pastebin) |
03:27.54 | baliktad | line 54, you have host = byo.engin.com.au |
03:28.15 | baliktad | but the call is coming from 203.161.164.69 |
03:28.35 | baliktad | argh, same |
03:30.39 | dramman | I'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) |
03:31.14 | baliktad | the 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) |
03:31.46 | baliktad | the [exgin_36_in] context provides all the details about handling a call that comes in |
03:31.55 | dramman | is [engin_36_in] a sip extension, or a context? |
03:32.23 | baliktad | s/context/account |
03:32.33 | baliktad | if it's in sip.conf, it's a SIP account |
03:32.38 | baliktad | you only have contexts in your dialplan |
03:32.57 | dramman | dialplan == extensions.conf? |
03:33.02 | baliktad | yes |
03:33.14 | drmessano | [Digg] Concept: Taking Social Networking To The Next Step <--- Face to face? |
03:33.51 | jaytee | Noooooo!!! I don't want to leave my house! |
03:33.52 | dramman | so what's the purpose of the last bit of register string after "/"? |
03:33.56 | drmessano | HAW |
03:33.57 | baliktad | have you reloaded your sip module and dialplan since making the changes you have pastebin'd? |
03:34.15 | drmessano | Who the hell wants to interact with live people? |
03:34.28 | baliktad | the last part tells asterisk what extension it should try and match when the call makes it to your dialplan |
03:34.29 | drmessano | writes off the whole idea |
03:34.30 | dramman | yes, continually doing "module reload" |
03:34.37 | jaytee | I've changed my mind, when I grow up I want to be a choo-choo instead of a firetruck |
03:35.02 | drmessano | lol |
03:35.24 | drmessano | I want a copy of Microsoft PBX Simulator 2007 |
03:35.32 | jaytee | or maybe a zamboni |
03:35.38 | jaytee | lol |
03:35.40 | Maliuta | jaytee: too much Vanilla Ice? decided you want to be a train and he'll be the caboose? ;) |
03:35.52 | drmessano | OMG |
03:35.55 | dramman | So by saying /0290115436 it'll call "extensions.conf::0290115436" which is essentially the same as [DID_engin_36_in]? |
03:36.07 | drmessano | Not so much the Vanilla Ice bust out.. but the reference |
03:36.25 | drmessano | Maliuta: The person using a Vanilla Ice reference on someone can't win |
03:36.26 | baliktad | first you need to get the call to match up to a SIP account |
03:36.34 | Maliuta | is frightened that he remembers VI lyrics |
03:36.36 | baliktad | it has to be matched to a sip peer for asterisk to accept the call |
03:36.39 | jaytee | drmessano, no one's gonna slide up to my bumper if I have anything to say about it |
03:36.48 | baliktad | once it does, it will find the context= line for that sip peer |
03:37.03 | drmessano | That's like "HA, you're just like that ABBA song" |
03:37.03 | Maliuta | drmessano: I win by default |
03:37.07 | drmessano | ^ Fail |
03:37.16 | dramman | so the problem lies in "sip.conf::[engin_36_in]"? |
03:37.20 | baliktad | then 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) |
03:37.24 | Maliuta | dramman: but _you're_ the dancing queen in here ;) |
03:37.38 | drmessano | tab FAIL |
03:37.39 | baliktad | yes, there is a problem matching to your sip peer |
03:37.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
03:37.52 | Maliuta | s/dramman/drmessano/ |
03:37.52 | baliktad | for some reason I can't yet figure out |
03:38.07 | drmessano | s/tab fail/HA! |
03:38.24 | drmessano | s/*/* |
03:38.32 | drmessano | omg a loop |
03:38.34 | Maliuta | reminds drmessano that he wins because he is "The Illegitimate Son of God" |
03:38.39 | jaytee | they taught us in class to make a sip peer account for our SIP provider and a sip user account to match |
03:39.58 | jaytee | Maliuta, who's the Illegitimate Son of God? you or drmessano? |
03:40.20 | Maliuta | jaytee: well obviously it can't be him |
03:41.02 | drmessano | Dad says to STFU |
03:41.12 | jaytee | reminds me of right before the election, I was getting all kinds of junk mail and fearmonger calls from the RNC slander/fearmonger unit. |
03:41.42 | Maliuta | KMFDM says "Spit Sperm" |
03:42.25 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
03:42.33 | jaytee | "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) |
03:43.16 | Maliuta | jaytee: who did you get to record that for your autodialer? |
03:43.45 | stencil | Ya, but Obama gives us cheap gas! |
03:43.52 | drmessano | Oh it was funny |
03:44.03 | jaytee | Allison wouldn't do it so I hired a dancer from P.T.'s Gentleman's Club |
03:44.08 | drmessano | I was reading about Paul Broun calling Obama a socialist, etc |
03:44.12 | baliktad | dramman any luck? |
03:44.19 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
03:44.20 | dramman | jaytee: : [engin_36_out] is my peer, [engin_36_in] is my user |
03:44.27 | drmessano | Hes this fucker from GA that kept sending us spam for WEEKS |
03:44.48 | drmessano | i am gonna e-mail him and ask for his resignation |
03:46.41 | harry_v | as long as obama has the right team then it will work. |
03:46.45 | Maliuta | that give Socialists a bad name |
03:47.05 | jaytee | dramman, then you need to put context="nameofcontext_to_handle_incoming_call" in your [engin_36_in] account |
03:47.31 | baliktad | ...he has that, line 52 |
03:48.28 | jaytee | supposedly 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. |
03:48.42 | baliktad | his incoming call isn't getting matched to any SIP account |
03:48.47 | baliktad | hahah brilliant |
03:49.02 | jaytee | the first part of that is true, the rest was humor |
03:49.11 | dramman | jaytee: yes, I have "context = DID_engin_36_in" |
03:49.19 | jaytee | at least that's what they said in class this week |
03:49.56 | jaytee | and is the context actually named DID_engin_36_in ? |
03:50.20 | drmessano | jaytee: You are my peer.. my base |
03:50.21 | dramman | yes |
03:50.43 | jaytee | drmessano, and you are the wind beneath my wings ;-) |
03:50.47 | harry_v | is there no zap show chanells in core? |
03:50.52 | drmessano | covers his butt |
03:50.57 | jaytee | lol |
03:51.07 | Maliuta | drmessano: here comes that train again |
03:51.23 | Maliuta | *cough*Ice Ice*cough* |
03:51.45 | jaytee | harry_v, if you type help zap at the CLI what do you see? |
03:51.47 | dramman | should I set nat, registersip...? |
03:51.50 | drmessano | jaytee: If you think about it.. Sounds like peers/users are going away and "friend" is becoming "peer" |
03:52.09 | jaytee | drmessano, that's kinda what I was thinking |
03:52.18 | harry_v | no such command |
03:52.22 | dramman | my outgoing calls are working |
03:52.34 | drmessano | A little sed and internal rewrite of friend to peer for deprecation |
03:52.36 | Maliuta | harry_v: have you loaded chan_zap? |
03:52.43 | jaytee | harry_v, did you compile your * install? |
03:52.44 | harry_v | compiled |
03:52.49 | harry_v | yes |
03:52.53 | jaytee | did you compile zaptel first? |
03:52.57 | harry_v | no |
03:53.00 | drmessano | ...... |
03:53.03 | drmessano | bingo! |
03:53.06 | jaytee | ah, wrong order pal |
03:53.06 | harry_v | hehe |
03:53.15 | jaytee | libpri, zaptel, asterisk |
03:53.16 | drmessano | You get to drink.... |
03:53.18 | drmessano | FROM THE FIREHOSE |
03:53.18 | harry_v | ohh gee, 3 years since last time i compiled it |
03:53.23 | baliktad | dramman can you paste the output of sip show peers |
03:53.45 | harry_v | so basicly recompile zap then asterisk |
03:53.50 | drmessano | harry_v: $5 fine for whining |
03:54.18 | harry_v | no its called recall everything from five years ago :) |
03:54.20 | jaytee | harry_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) |
03:54.36 | harry_v | I know jaytee :) |
03:54.48 | jaytee | note the double ampersand there, or you could break it out into separate commands |
03:54.54 | dramman | http://pastebin.com/d68724f7e |
03:55.00 | jaytee | actually might want to do a make clean first |
03:55.20 | baliktad | ok, it's as I expected |
03:55.45 | jaytee | engin_36_in isn't registering |
03:56.03 | baliktad | your engin_36_in isn't being processed as a peer |
03:56.44 | *** part/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
03:56.56 | dramman | it's set to "type=user" |
03:57.23 | baliktad | if 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 |
03:57.47 | harry_v | right now though, my real issue is vm not kicking in the end of the 25 second ring duration. |
03:58.13 | baliktad | doh |
03:58.37 | baliktad | I always get confused too |
03:58.46 | harry_v | ohh wow, for some off reason its adding a module |
03:59.18 | harry_v | probebly do do with the zaptel needing sounds. |
04:00.06 | baliktad | if you want to accept calls from someone, you need to have an account as type peer or friend |
04:01.00 | dramman | I thought peer was for making calls |
04:01.22 | dramman | Have I got peer/user swapped? |
04:02.12 | dramman | I've not been able to find documentation which makes it very clear - it usually just lists the options and assumes that you know |
04:02.23 | jaytee | dramman |
04:02.30 | jaytee | did you look in the book? |
04:02.40 | jaytee | ~book |
04:02.41 | jbot | hmm... 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 |
04:03.32 | harry_v | jaytee, thanks for the reminder. btw, what version are you running? |
04:03.36 | Maliuta | we should train him to respond to ~worshiphere |
04:03.52 | *** join/#asterisk ManxPower (n=manxpowe@55.sub-75-202-121.myvzw.com) |
04:06.01 | ManxPower | registration 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. |
04:06.09 | jaytee | harry_v, depends which server. I've got 1.4.15 and 1.4.22 in production and 1.6.0.2 in test |
04:06.58 | harry_v | Manx, am i correct that 1.4.X did away with n+101 for the number sequence in a dial pattern for extentions.conf? |
04:09.31 | jaytee | n+101 was priority jumping, not dialed number manipulation |
04:09.41 | harry_v | yes |
04:09.51 | harry_v | that was what i was refering to |
04:10.11 | ManxPower | harry_v: I assume so but the official document is called...can you guess? |
04:10.17 | jaytee | it still works in 1.4 but it was listed as deprecated and you should use labeled priorities instead. |
04:10.32 | jaytee | NOBODYREADSME.TXT |
04:10.38 | ManxPower | jaytee: nope. |
04:10.45 | harry_v | I 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. |
04:10.46 | jaytee | UPGRADE.TXT? |
04:11.03 | jaytee | or UPGRADE1.2-1.4.txt |
04:11.06 | ManxPower | UPGRADE.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) |
04:11.48 | jaytee | I think I've seen those files in a folder somewhere. wonder what's in 'em? :-) |
04:12.05 | ManxPower | jaytee: nothing important, I'm sure! |
04:12.38 | ManxPower | I 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) |
04:14.30 | harry_v | i will have to troubelshoot this latter. |
04:14.34 | jaytee | ManxPower, 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. |
04:15.01 | ManxPower | jaytee: these youngsters don't know how good they have it! |
04:16.07 | jaytee | nope, 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. |
04:16.10 | ManxPower | jaytee: Some people think Asterisk does not have good docs -- try the Novatel Wireless (CDMA) Toolkit and you'll see what bad docs are. |
04:17.02 | ManxPower | They actually FORGOT to include the sample source code that is referenced all over the SDK in the Linux build of the SDK. |
04:17.21 | jaytee | ManxPower, my Digium backpack had a copy of AsteriskNOW for Dummies that comes with the software on DVD. Must be a consolation prize :-) |
04:17.47 | Maliuta | they're giving you coasters? |
04:18.03 | ManxPower | AsteriskNOW for Dummies? Isn't that like "Eating Celery for Dummies"? |
04:18.18 | jaytee | Maliuta, no. Coasters come from AOL |
04:18.37 | Maliuta | or encouraging you to engage in team work and build a barometer? |
04:18.48 | jaytee | Celery, 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) |
04:20.13 | Maliuta | jaytee: you never got exposed to MSDCD's? |
04:20.22 | drmessano | 1.6.0.2 is out? |
04:20.28 | Maliuta | I had literally close to a thousand |
04:20.43 | Maliuta | drmessano: it might be in the closet |
04:20.55 | drmessano | closet? |
04:20.57 | drmessano | oh |
04:21.01 | drmessano | ...... |
04:21.05 | Maliuta | drmessano: 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) |
04:22.24 | jaytee | oh, sorry. that was a typo. 1.6.0.1 |
04:22.25 | dramman | I've commented out [engin_36_in] and am trying to do everything with [engin_36_out] (will rename later if it works) |
04:22.36 | dramman | sip debug still says " |
04:22.59 | dramman | no matching peer or user for '203.161.160.69:5060' |
04:25.11 | dramman | (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) |
04:25.24 | dramman | NOTICE[3967]: chan_sip.c:13885 handle_request_invite: Call from '0290115436' to extension 'DID_engin_36_in' rejected because extension not found. |
04:27.15 | jaytee | how does someone dial DID_engin_36_in from a phone? they didn't teach us that in class. |
04:27.18 | Maliuta | dramman: next question do you have an exten => 0290115436 in that context |
04:27.49 | ManxPower | jaytee: his register statemnet is prolly screwed up |
04:28.05 | Maliuta | jaytee: how do you dial SIP/nikolai@voip.kissmyass.com from a phone? |
04:28.35 | Maliuta | :) |
04:28.56 | dramman | In extensions.conf::[DID_engin_36_in] I've got "exten => s,1,NoOp(Call from Engin)" "exten => s,n,Dial(SIP/7000,20) |
04:29.28 | ManxPower | dramman: "s" means "the technology is too stupid to send us a dialed number", usually FXO ports. |
04:29.37 | Maliuta | it's not going to 's' |
04:30.02 | Maliuta | ManxPower: no, my ITSP sends to s |
04:30.16 | ManxPower | Maliuta: then your ITSP is too stupid. |
04:30.29 | dramman | ...now, there isn't actually a [7000] defined in sip.conf, but there is a "exten => 7000,1,Goto()" in extensions.conf::[default] |
04:30.37 | Maliuta | I guess they expect you'll have just on device or on context per account |
04:30.50 | dramman | Isn't "s" |
04:30.54 | dramman | everything? |
04:31.08 | ManxPower | no, "s" means "NOTHING" |
04:31.17 | ManxPower | "_." means "everything" |
04:31.43 | jaytee | or s means "um, I guess I'd better start here since they didn't give a number to dial" |
04:31.50 | drmessano | s = stupid _. = "No period, oh shit" |
04:32.33 | ManxPower | I assume ATFOT talks about "s" |
04:32.55 | jaytee | ManxPower, careful with the assume, buddy! |
04:33.14 | Maliuta | ManxPower: IIRC that's a diaplan for and FXO |
04:34.34 | jaytee | wow, check this out!!! it's awesome!!! http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
04:34.57 | ManxPower | If your DID is 5045551212 then usually you would exten => 5045551212,1,Dial(whatever |
04:35.20 | drmessano | You know what happens when you assume |
04:35.26 | drmessano | You make an ass out of Maliuta |
04:35.33 | drmessano | ducks |
04:35.51 | Maliuta | kicks drmessanos ducks in a row for him |
04:36.32 | drmessano | throws a handful of fair dinkum |
04:36.56 | *** join/#asterisk hadronzoo (n=user@gateway.publicvpn.net) |
04:37.35 | Maliuta | kicks drmessano innanuts |
04:38.03 | drmessano | yep, you got me there |
04:38.11 | ManxPower | jaytee: what did you think of the class? |
04:38.15 | jaytee | man, this part of Huntsville is boring |
04:38.28 | drmessano | When do you go back? |
04:38.35 | dramman | Ok, I've swapped the "s" with "_." and it's trying to dial SIP/7000 - not exactly working, but a lot better |
04:38.46 | jaytee | ManxPower the class was awesome, right up till the point where I ran out of time doing the practical lab and scored a 45 |
04:38.57 | jaytee | I fly out tomorrow |
04:39.12 | ManxPower | dramman: unfortunately you cannot just take bits and pieces and expect it to work as expected. Go read the Asterisk Book. |
04:39.38 | ManxPower | because with _. you will have dialplan loops as the same extension will match 12345 and match "h" and "i" |
04:39.50 | drmessano | I tried to cut and paste out of the book and now my monitor wont shut off.. damn Elmers glue |
04:39.55 | ManxPower | So go read the book and save yourself weeks of problems. |
04:40.16 | jaytee | personally 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. |
04:40.32 | ManxPower | You almost NEVER want to "match all" In fact the CLI should complain if you use _. |
04:40.33 | Maliuta | Go read the book and save the regulars from hunting you down and beating you to death with it |
04:40.52 | drmessano | jaytee: Where you screwed up is you forgot your years of watching 80s TV shows |
04:41.24 | ManxPower | jaytee: Did outside/voip calls sound crappy? |
04:41.37 | drmessano | jaytee: 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 |
04:41.40 | jaytee | well, I had the option to not use and configure my T1 card but I foolishly thought "what the hell, I've done this before" |
04:42.10 | Maliuta | is thankful the tv show he is watching right now is '70s not '80s |
04:42.12 | drmessano | Because in the 80s, you would have gotten away with it |
04:42.22 | jaytee | ManxPower, in the class we were just emulating outside calls, not actually going outside |
04:42.45 | drmessano | and your teacher with the unusually large glasses would have had to accept it and look stupid |
04:42.48 | ManxPower | in the fast start we actually made some outside calls. |
04:43.03 | ManxPower | sounded like trying to send audio thru a broadcast storm |
04:43.05 | jaytee | drmessano, Jared doesn't wear glasses |
04:43.26 | drmessano | jaytee: He would if this was growing pains or Family Ties |
04:43.31 | jaytee | lol |
04:43.41 | drmessano | BUT |
04:44.08 | drmessano | After all that coolness, you would have had to confront skippy or boner about their drug problem |
04:44.09 | ManxPower | I'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. |
04:44.24 | jaytee | this was only the 3rd Advanced class, they're still fine tuning the curriculum |
04:44.39 | ManxPower | *nod* |
04:44.55 | Maliuta | jaytee: anyone interesting running them? |
04:45.11 | drmessano | jaytee: to put that into context here, after "cooly outdoing" Jared, you would have to talk to Russell about his blackberry addiction |
04:45.13 | jaytee | Maliuta, running what? |
04:45.14 | dramman | My config files are basically taken from the asterisk samples and the book. On page 127 it defines [incoming] exten => s,1,Answer() |
04:45.29 | Maliuta | jaytee: the classes |
04:45.38 | drmessano | To which Russell would have replied "I dont have a problem dude, dont be downer" |
04:45.41 | drmessano | To which Russell would have replied "I dont have a problem dude, dont be A downer" |
04:45.49 | ManxPower | dramman: The asterisk .sample files are designed to show as many options as possible, not to actually work. |
04:45.53 | drmessano | watched too much 80s TV |
04:46.09 | jaytee | 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 |
04:46.22 | Maliuta | passes drmessano a pile of CHiPS episodes to watch |
04:46.26 | drmessano | Oh god |
04:46.31 | drmessano | Now Jaytee is gonna be a name dropper |
04:46.38 | drmessano | Nice knowing you, slick |
04:46.58 | jaytee | drmessano, he asked. I didn't volunteer it |
04:47.08 | drmessano | Im just screwing with ya |
04:47.21 | Maliuta | I can try an out drop him, but I think mine are fairly run-of-the-mill names |
04:47.28 | drmessano | Besides, you didnt elaborate enough for a full, egomaniacal namedrop |
04:47.28 | jaytee | but when we were having lunch with Mark...... :-) |
04:47.41 | Maliuta | Suter was there? |
04:47.42 | drmessano | Let me fix your sentence |
04:47.56 | jaytee | who's Suter? |
04:48.06 | Maliuta | I haven't seen Mark Suter in ages, I thought he was still in Canberra |
04:48.16 | Maliuta | taps his nose |
04:48.48 | jaytee | never heard of him. is he a notable name in the * "community"? |
04:49.11 | Maliuta | no, but he is a notable name in these parts |
04:49.23 | Maliuta | google is your friend |
04:49.31 | drmessano | "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 :)" |
04:49.33 | drmessano | There |
04:49.37 | drmessano | Thats about right |
04:49.39 | jaytee | throws Maliuta a Scooby Snack |
04:50.45 | jaytee | wiseass! :-) |
04:51.37 | drmessano | "I have been using Asterisk since 0.0.9. Notice the earliest posted on the download site is 0.1.0? ;)" |
04:51.43 | drmessano | I think thats FTW |
04:52.02 | Maliuta | back when I was trading patches with W Richard Stevens ..... |
04:52.21 | drmessano | Oh god, that reminds me of this time stallman and I... |
04:52.43 | jaytee | I 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 |
04:53.27 | Maliuta | I 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 |
04:53.27 | drmessano | No, I am really Kerry Garrison |
04:53.36 | drmessano | Now kiss my ass.. and give me back my camera |
04:53.45 | jaytee | Maliuta, who? russell? |
04:55.31 | file | wobbles |
04:55.33 | jaytee | both 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. |
04:55.53 | jaytee | weebles wobble but they don't fall down! |
04:55.55 | Qwell | jaytee: developers don't actually USE the stuff |
04:55.56 | Maliuta | jaytee: W Richard Stevens |
04:56.04 | file | I use it! |
04:56.45 | jaytee | yes, you do! |
04:56.54 | jaytee | and I for one am grateful for that |
04:57.02 | ManxPower | I hink my TiVo is freaking out. |
04:57.05 | drmessano | Qwell: You dont need to tell us the developers dont use this effing thing.. The code wreaks of it. |
04:57.15 | drmessano | ducks |
04:57.20 | jaytee | file, and thanks for the tip on preloading grammars by the way |
04:57.36 | file | jaytee: working well? |
04:57.47 | *** join/#asterisk styelz (n=yoohoo@egg.vividas.com) |
04:57.59 | jaytee | Qwell, ignore him. Kerry compiled the extra-snarky module when he created the drmessano bot :-) |
04:58.37 | jaytee | file, yes. each section of the IVR no longer has a slight lag like it did. |
04:58.57 | file | jaytee: good |
04:59.18 | drmessano | I 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" |
04:59.24 | drmessano | Bah humbug |
04:59.53 | drmessano | heh |
04:59.53 | jaytee | I was working on it over VPN today and adding some other tweaks to it I picked up from the class |
05:00.06 | drmessano | Actually, I owe file a beer |
05:00.16 | file | I don't normally drink beer |
05:00.31 | jaytee | what's your favorite poison? |
05:01.08 | drmessano | I got G726 running with one of my providers.. had to use g726aal2, and actually understood all the cryptic BS about G726 thanks to him |
05:01.09 | file | purple haze martini, or a good strawberry daiquiri |
05:01.10 | *** join/#asterisk Kdas (n=Kdas@c-98-207-95-143.hsd1.ca.comcast.net) |
05:01.16 | Kdas | wass up ???? |
05:01.16 | drmessano | Worked beautifully |
05:01.46 | Kdas | anyone know why voicemails on my asterisk box don't alert my handytone286? |
05:01.53 | drmessano | Genetics |
05:02.09 | jaytee | what's in a purple haze? I've had a pomegranate vodka martini and it was tasty |
05:02.25 | jaytee | hahaha, genetics |
05:02.42 | file | jaytee: pretty much that |
05:03.08 | jaytee | ah, never heard it called a purple haze. you're in Toronto, right? |
05:03.16 | file | Moncton, NB |
05:03.43 | drmessano | Pretty much the answer to "Why doesn't my grandstream ______" is "It's a grandstream" |
05:03.44 | jaytee | wow, up on the far northeastern tip of the continent then. |
05:04.16 | file | jaytee: yup |
05:04.21 | jaytee | "I can see Greenland from my house!" |
05:04.55 | drmessano | Its 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 |
05:05.13 | drmessano | If you listen closely during a call, you can hear pole position |
05:05.14 | jaytee | hahahaaha |
05:05.23 | jaytee | rofl |
05:05.38 | Kdas | so... anyone know why i am not geting voicemail alerts on my ht286 ? |
05:05.49 | drmessano | "because it's a grandstream" |
05:05.54 | drmessano | ~grandstream |
05:05.54 | jbot | grandstream is, like, the Yugo of VoIP hardware. Run. Run away now. |
05:06.00 | *** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
05:06.05 | echelon | hi! :) |
05:06.08 | Kdas | :(( |
05:06.12 | echelon | what's a good ATA? |
05:06.16 | coppice | grandstream 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 |
05:06.17 | echelon | for regular phone |
05:06.19 | drmessano | not a grandstream |
05:06.28 | jaytee | linksys PAP2-T |
05:06.45 | echelon | hmm.. i've been hearing that as well |
05:06.54 | echelon | it supports fax? |
05:06.56 | Kdas | so i am screwed ? |
05:06.57 | jaytee | or if you want a little more bang for your buck, SPA2102 |
05:07.09 | coppice | PAP2T == good ATA if you don't use FAX |
05:07.11 | coppice | SPA2102 == good ATA if you do use FAX |
05:07.27 | echelon | it has FXO though? |
05:07.35 | drmessano | No |
05:07.41 | drmessano | Who said FXO? |
05:07.56 | coppice | SPA3102 is like an SPA2102 but one of the ports is changed to FXO |
05:08.29 | jaytee | yeah, the 3102 has a router and FXO |
05:08.37 | jaytee | and an FXS port |
05:08.48 | jaytee | the 2102 is just 2 FXS ports |
05:09.35 | jaytee | Kdas, do you have a mailbox= statement in the sip.conf for that user? |
05:09.44 | Kdas | jaytee, yes sir |
05:09.52 | echelon | oi.. $63 |
05:10.21 | jaytee | lose the sir, that's like putting a chandelier in an outhouse |
05:11.04 | jaytee | Kdas, been ages since I setup an HT286 |
05:11.19 | jaytee | can't remember if there were any MWI options. |
05:11.24 | Kdas | jaytee, mwi ? |
05:11.56 | *** join/#asterisk Vco (n=Vco@S0106000db912f754.cg.shawcable.net) |
05:12.07 | jaytee | Message 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 |
05:12.18 | Kdas | jaytee, oh wait a second i have a mailbox= on my [callwithus] not under my [phone] |
05:12.40 | Kdas | jaytee, yea ht286 subosibly supports that |
05:13.54 | jaytee | Kdas, you need to add the mailbox="mailbox#"@default (or some other context if you've setup different ones in voicemail.conf |
05:14.27 | Kdas | jaytee, do they go under the phone or my voip provider contexT? |
05:14.37 | jaytee | in 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. |
05:15.11 | Kdas | sorry i didn't understand that |
05:15.12 | jaytee | Kdas, the phone |
05:15.24 | Kdas | jaytee, ok i set that i will test that |
05:15.39 | jaytee | in your sip.conf file you have a user set to the ATA, correct? because it's SIP to the ATA. |
05:16.17 | Kdas | jaytee, yes sir |
05:16.27 | jaytee | your handydandytone registers to * as a sip peer. |
05:17.01 | jaytee | and you setup a user account in sip.conf for that peer and that's where you set the mailbox= statement. |
05:17.09 | jaytee | Kdas, have you read the book much? |
05:17.12 | jaytee | ~book |
05:17.13 | jbot | well, 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 |
05:17.41 | Kdas | jaytee, yea i just been gone for a while till i got sick of not geting a voicemail flash |
05:18.26 | Kdas | jaytee, ok i get it so my handytone is blinking but my phone dosent seem to be getting it |
05:19.13 | jaytee | use the web interface on the handydandytone to check if there's a stutter dialtone option for MWI |
05:21.28 | Kdas | ok well right now i think its updating firmware but thatnks for help ;) |
05:22.14 | jaytee | man, 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. |
05:22.35 | Kdas | haha |
05:23.29 | jaytee | I can get direct flights to just about anywhere from Indianapolis except for Huntsville. |
05:26.22 | jaytee | I hate layovers at O'Hare. I always feel like Tom Hanks in that movie where he had to live at the airport |
05:26.51 | harry_v | hehe |
05:27.07 | harry_v | That was a pretty funny movie. |
05:29.27 | harry_v | well, its not as bad as taking a military HOP |
05:29.39 | harry_v | But at least it is a free ride. |
05:30.56 | drmessano | Your in Canada, right? |
05:31.01 | drmessano | you're |
05:31.26 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
05:32.16 | jaytee | who's in Canada? |
05:32.21 | drmessano | harry_v |
05:32.24 | jaytee | ah |
05:32.29 | drmessano | He said "Military" |
05:32.39 | drmessano | I was confused.. didn't know canada had one |
05:32.55 | jaytee | never heard of the RAF? |
05:33.21 | drmessano | Is that like the RCMP? |
05:33.24 | jaytee | I'm trying to remember their air force stunt team's name |
05:33.50 | jaytee | like the Blue Angels or the USAF Thunderbirds. They put on a hell of a show. |
05:33.58 | drmessano | I seriously doubt canada has an air force.. Where the hell would they put the horse... |
05:33.59 | harry_v | yes |
05:34.15 | harry_v | dr, im a US resident living up here. |
05:34.29 | drmessano | In Vancouver? |
05:34.42 | harry_v | Canada has a Airfoce. Every heard of Canadian F-18's? |
05:34.52 | harry_v | In the Vancouver area |
05:34.55 | jaytee | drmessano, hehe, I'm former USAF and I worked with some Canadian RAF people in the past and they were top notch professionals. |
05:34.56 | drmessano | OMG if you see Richard Dean Anderson, tell him I SO LOVE MACGYVER ZOMG 4EVA |
05:35.13 | jaytee | rofl |
05:35.21 | harry_v | jaytee, what was your AFSC? |
05:35.30 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
05:35.35 | harry_v | yea, he is cool |
05:35.36 | jaytee | harry_v, 30750 |
05:35.41 | drmessano | I stopped flying jets after I got shot down over Kumbang |
05:35.50 | harry_v | :) |
05:36.20 | harry_v | I worked on the Jollies |
05:36.21 | jaytee | he still comes in here once in awhile looking for drmessano |
05:36.35 | Juggie | jaytee, blue birds |
05:37.03 | drmessano | I was 20 clicks north of Hunglad and was taking heavy fire from Tong-Po and most of General Tso's army |
05:37.06 | harry_v | Gone are the days of working on the flightline and dealing with weather conditions |
05:37.09 | jaytee | Juggie, the precision flying team? that sounds right |
05:37.28 | harry_v | I imagine nam was pretty bad |
05:37.32 | jaytee | Jollies? the Jolly Greens? those big ass dual rotor helos? |
05:37.35 | harry_v | yes |
05:37.43 | harry_v | 3/53's |
05:37.48 | Juggie | jaytee, that is right :) |
05:37.52 | harry_v | dual=chinook |
05:37.53 | jaytee | ah, in the USAF? or another branch? |
05:37.57 | harry_v | usaf |
05:38.01 | harry_v | SpecialOps |
05:38.06 | jaytee | :-) |
05:38.11 | drmessano | harry_v: That wasn't vietnam, that was the buffet line at the Ming-Yat restaurant.. but almost as traumatic |
05:38.27 | jaytee | harry_v, Pararescue? |
05:38.32 | harry_v | rotorhead |
05:38.43 | drmessano | rotorhead? |
05:38.47 | drmessano | Propellorhead |
05:38.49 | harry_v | crew chief |
05:39.07 | Juggie | ah |
05:39.14 | jaytee | my dad was a crew chief on B-17's in the 8th in England during WWII |
05:39.15 | Juggie | jaytee, i was close |
05:39.17 | Juggie | but not right |
05:39.18 | Juggie | snowbirds |
05:39.21 | Juggie | not bluebirds |
05:39.29 | harry_v | That was how i started my career. Still have a strong interest in anything mechanical. |
05:39.30 | drmessano | My dad was kicked out of Vietnam |
05:39.31 | jaytee | Juggie, YES!!! that's it. thanks dude |
05:39.37 | drmessano | Thats his claim to fame |
05:39.41 | Juggie | http://www.snowbirds.dnd.ca/site/index_e.asp |
05:39.55 | harry_v | If I had the money and a shop would not mind buying a T-58-GE-5 Turbo shaft jet engine. |
05:40.03 | harry_v | Or some other variant |
05:40.08 | jaytee | harry_v, my dad's nickname growing up was Tinker. he loved taking things apart and putting them back together. |
05:40.23 | jaytee | he was an excellent mechanic |
05:40.25 | drmessano | My 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 |
05:40.30 | harry_v | :) I was one of those. But money was a issue. |
05:40.39 | harry_v | hehe |
05:40.40 | jaytee | and guess what base I got stationed at? |
05:40.50 | harry_v | Reklavic? |
05:40.52 | harry_v | :) |
05:41.06 | harry_v | or some horrid base on the Alutian Chains |
05:41.07 | jaytee | no, Tinker AFB in Oklahoma City |
05:41.12 | harry_v | ohh |
05:41.19 | harry_v | man that area is flat |
05:41.21 | jaytee | yep |
05:41.34 | harry_v | I was at Sheppard tx so I know :) |
05:41.49 | harry_v | kinda strange to see a city 20 miles away. |
05:41.49 | jaytee | I've been to Cheyenne Mountain in Colorado Springs and Elmendorf AFB in Alaska |
05:41.55 | harry_v | nifty |
05:42.11 | harry_v | how many doors do thay have in that mountain? |
05:42.16 | drmessano | I remember when I was at Cheyenne mountain |
05:42.23 | drmessano | Playing tic tac toe with the WOPR |
05:42.26 | drmessano | Good times |
05:42.36 | harry_v | your kidding right? |
05:42.39 | jaytee | I didn't count em. I saw about 4 huge blast doors |
05:43.05 | drmessano | well, they called me in because they had a problem |
05:43.16 | harry_v | i see |
05:43.17 | jaytee | but there might have been more. I only got to go to certain areas inside. |
05:43.24 | harry_v | right |
05:43.26 | drmessano | See, the computer thought it was playing Globalthermonuclear War |
05:43.37 | drmessano | But it was a simulation |
05:44.10 | drmessano | So 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 |
05:44.17 | harry_v | I thought this was of movie making game playing. your kidding right? |
05:44.45 | jaytee | my 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. |
05:44.59 | harry_v | my base was perhaps one of two that stored all the nucks. |
05:45.06 | harry_v | But i wont say where it was :) |
05:45.31 | drmessano | Well, the WOPR had a few bugs |
05:45.43 | drmessano | I blame Mr McKittrick |
05:45.46 | harry_v | I really enjoyed the movie |
05:45.53 | jaytee | harry_v, I bet i could guess :-) |
05:45.55 | drmessano | He looked a lot like Dabney Coleman |
05:46.37 | drmessano | After the computer guessed the launch codes, man, I thought we were toast |
05:46.44 | harry_v | :) |
05:46.47 | drmessano | CPE1704TKS it read |
05:46.52 | jaytee | harry_v, New Mexico? |
05:46.56 | drmessano | Talk about jaws dropping |
05:47.34 | drmessano | But man, when that guy from Alaska was like "Sir, um sir, we're still here" |
05:47.42 | drmessano | That... was win |
05:47.55 | harry_v | I thought that was great |
05:48.21 | harry_v | such a good movie of its time. Dont forget the made for TV movie "The day after" |
05:48.39 | drmessano | Um.. I wasnt describing a movie |
05:48.44 | harry_v | with todays solid state world, EMP is still a serios threaght. |
05:50.05 | drmessano | EMP is actually counterproductive |
05:50.06 | harry_v | yea, 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. |
05:50.10 | jaytee | it's even more a threat today than it was in the 70's and 80's |
05:50.11 | harry_v | how so |
05:50.14 | drmessano | i doubt it will ever be a real threat |
05:50.35 | drmessano | Well, shenanigans you plan to benefit from using EMP will be destroyed by EMP |
05:51.03 | harry_v | my truck in general, will survive |
05:51.24 | drmessano | Youre missing the big picture |
05:51.42 | jaytee | one 20 megaton warhead detonated 80 to 90 miles up over Kansas City could fry most of the electronics in the lower 48 |
05:52.10 | harry_v | jaytee, was that from actuall DOD material? |
05:52.13 | drmessano | War is useless without the raping and pillaging |
05:52.34 | echelon | how do services like ipkall and grandcentral get all those phone numbers? |
05:52.44 | jaytee | harry_v, of course not! I'd never divulge classified information |
05:52.50 | harry_v | heheh |
05:52.55 | jaytee | echelon, they steal them |
05:52.57 | harry_v | I know |
05:52.59 | drmessano | There will be nothing to pillage with EMP.. No electronic transfers, no real goods |
05:53.13 | echelon | seriously |
05:53.17 | drmessano | Just damage... which will be effective, but have less of a point |
05:53.24 | drmessano | echelon: They buy them |
05:53.42 | echelon | drmessano: just a one time fee? |
05:53.48 | drmessano | No |
05:53.51 | harry_v | thats why emergency supplies like food and fuel are good things to have on hand. |
05:54.00 | echelon | drmessano: so how can they provide them for free? |
05:54.07 | drmessano | They make money from them |
05:54.12 | echelon | how? |
05:54.16 | harry_v | in fact, food would have been a good investment after the stock crash if you have the storage to hold it. |
05:54.19 | drmessano | They're not robin hood |
05:54.23 | jaytee | if 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 |
05:54.37 | drmessano | from collecting intercarrier transfer fees for terminated calls |
05:55.09 | coppice | jaytee: rubbish. the human race is far from having the capability to cauterise the planet |
05:55.15 | echelon | oh |
05:55.19 | drmessano | harry_v: Youre missing the whole point |
05:55.24 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-158-157.lns10.mel4.internode.on.net) |
05:55.41 | harry_v | i know alot more supplies would be needed. |
05:55.42 | echelon | drmessano: does the person receiving the calls get charged? or just their carrier? |
05:55.47 | drmessano | harry_v: I am talking about MOTIVATION, not HOW GOOD YOUR TRUCK IS |
05:56.26 | drmessano | There'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 |
05:56.48 | harry_v | Explain motivation ? |
05:56.51 | jaytee | I'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. |
05:56.55 | drmessano | I just did |
05:57.04 | *** part/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
05:57.08 | drmessano | Ok, let me make this easy for you |
05:57.09 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
05:57.49 | drmessano | If 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 |
05:58.02 | harry_v | exactly |
05:58.15 | drmessano | We destroy the worldly assests of someone electronically, and we lose all their worldly assets we came to STEAL |
05:58.21 | harry_v | just nuke the area with a nutron bomb. |
05:58.23 | drmessano | So EMP becomes STUPUD |
05:58.26 | drmessano | STUPID |
05:58.30 | drmessano | .... |
05:58.32 | drmessano | WTF dude |
05:58.42 | harry_v | kills everyone and does not destry the infrastructor |
05:59.21 | drmessano | Um yes, it does |
05:59.27 | jaytee | better to use a trojan worm that just transfers all their ones and zeroes to an undisclosed account in the Cayman Islands |
05:59.47 | jaytee | and then nuke em! |
06:00.27 | harry_v | I remember in my training I had to learn about the nuclear triad. |
06:01.01 | [TK]D-Fender | jaytee: Which movie? |
06:01.03 | harry_v | then chemical training. |
06:01.14 | drmessano | You 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 |
06:01.20 | drmessano | Thats very, very stupid |
06:01.36 | [TK]D-Fender | jaytee: She was in BloodRayne which was a total piece of garbage |
06:02.11 | harry_v | anything underground though, would be protected. |
06:02.25 | drmessano | Yeah.... about that |
06:02.26 | harry_v | or in a feraday cage |
06:02.30 | drmessano | Everything overground would not be |
06:03.16 | harry_v | most usaf and who what other millitary aircraft are hardened against a emp attack. |
06:03.35 | harry_v | Anywa on to another subject |
06:03.52 | drmessano | Yes, please |
06:04.31 | drmessano | Before we start talking about the best way to recover a fumble in a football game is by blowing up the actual football |
06:04.40 | drmessano | duh |
06:05.28 | harry_v | probebly more low level localized emp attack would be effective |
06:08.55 | coppice | EMP's da bomb! |
06:09.03 | drmessano | lol |
06:10.10 | drmessano | That reminds of a line.. |
06:10.17 | drmessano | That reminds me of a line.. |
06:10.27 | drmessano | Something about the horse you rode in on.. |
06:10.30 | drmessano | Oh wait, ha |
06:10.46 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
06:11.05 | drmessano | So whats why we HATE australia |
06:11.11 | drmessano | oh heh, hi Maliuta |
06:11.17 | drmessano | thats* |
06:11.37 | drmessano | too.... crosseyed... |
06:12.03 | drmessano | jaytee: I loaded 1.6.1 on another box, it failed miserably as well |
06:14.59 | *** join/#asterisk TrentCreek (n=kvirc@adsl-66-138-255-185.dsl.hrlntx.swbell.net) |
06:15.38 | [TK]D-Fender | drmessano: 1.6.1 from beta, or 1.6.0.1? |
06:16.07 | *** join/#asterisk kisu (n=hexago@daniel1117.broker.freenet6.net) |
06:19.34 | *** part/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
06:19.57 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
06:20.03 | jaytee | drmessano, you could always try using Subversion and loading from the latest from trunk instead of one of the branches :-) |
06:21.09 | jaytee | maybe by some miracle they've got it fixed already in trunk |
06:21.41 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
06:21.49 | [TK]D-Fender | jaytee: So which movie was it? |
06:22.16 | jaytee | [TK]D-Fender, that's on now? |
06:22.36 | jaytee | Bloodrayne |
06:22.47 | *** join/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
06:22.49 | TrentCreek | anyone use FreePBX with asterisk2billingg? |
06:22.55 | jaytee | she's the one that played the new terminator in T3 |
06:23.52 | [TK]D-Fender | jaytee: Yeah, it was craptastic |
06:24.05 | jaytee | I 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-Fender | jaytee: She was smoking hot in Martal Kobat:The Series which is what launched her |
06:24.37 | [TK]D-Fender | jaytee: the Deckard question is a very open-ended point. Thats the best part |
06:25.01 | drmessano | Oh god, spoilers |
06:25.09 | drmessano | Hogwarts was really an STD |
06:26.33 | jaytee | lol |
06:26.56 | jaytee | I love this one tshirt they had on tshirthell.com "Dumbledore dies on page 573" |
06:27.37 | drmessano | Mark Spencer accidentally invented asterisk when he compiled apache in the same directory as teamspeak |
06:27.42 | jaytee | Phillip K. Dick was an awesome author |
06:27.48 | jaytee | lol |
06:29.17 | coppice | "Slavery gets shit done" :-) |
06:35.07 | jaytee | well, I'm gonna try and get some sleep. got a flight to catch in the morning |
06:35.12 | jaytee | later all |
06:47.03 | *** join/#asterisk feeds (n=feeds@85-135-225-22.adsl.slovanet.sk) |
07:17.01 | *** join/#asterisk korihor (n=korihor@200-71-160-1.genericrev.telcel.net.ve) |
07:31.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
07:47.23 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-37-217.w86-215.abo.wanadoo.fr) |
07:47.28 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
07:53.03 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
08:03.04 | *** part/#asterisk korihor (n=korihor@200-71-160-1.genericrev.telcel.net.ve) |
08:24.21 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-1-196.dsl.hrlntx.sbcglobal.net) |
08:48.14 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
09:05.04 | *** join/#asterisk Kdas (n=Kdas@c-98-207-95-143.hsd1.ca.comcast.net) |
09:05.20 | Kdas | hey all |
09:05.46 | Kdas | i am getting "Apr 13 22:03:35 NOTICE[2553]: chan_sip.c:5473 sip_reg_timeout: — Registration for '02490XXXX@nexcom.bg' timed out, trying again (Attempt #2)" i already have srvlookup=yes in sip.conf so whats wrong ? |
09:05.51 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
09:10.09 | Kdas | anyone ? |
09:12.44 | TrentCreek | it means what it means |
09:14.17 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
09:16.30 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
09:17.06 | Kdas | os everyone dead ? |
09:17.31 | Kdas | s/os/is |
09:20.53 | *** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
09:20.54 | echelon | hi |
09:23.54 | *** join/#asterisk Kdas (n=Kdas@c-98-207-95-143.hsd1.ca.comcast.net) |
09:24.01 | Kdas | umm |
09:25.41 | echelon | i'm thinking about purchasing this.. http://www.grandstream.com/ht286.html |
09:26.17 | echelon | after i enter all the sip credentials, i can just dial any ptsn # as i would with a regular line? |
09:30.44 | Kdas | i am getting "Apr 13 22:03:35 NOTICE[2553]: chan_sip.c:5473 sip_reg_timeout: — Registration for '02490XXXX@nexcom.bg' timed out, trying again (Attempt #2)" i already have srvlookup=yes in sip.conf so whats wrong ? |
09:38.15 | TrentCreek | JUST WHAT IT SAYS |
09:38.44 | TrentCreek | seems to be it's got a problem logging in |
09:38.52 | TrentCreek | or a problem with that domain name |
09:38.59 | TrentCreek | or could be a firewall problem |
09:40.20 | TrentCreek | try turning on SIP debug |
09:45.30 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
09:56.46 | *** part/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
10:21.20 | *** join/#asterisk ggiusti (n=giovanni@85-18-194-15.ip.fastwebnet.it) |
10:23.25 | *** join/#asterisk feeds (n=feeds@85-135-225-22.adsl.slovanet.sk) |
10:49.39 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
10:56.54 | TrentCreek | nope |
11:05.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
11:25.53 | hi365 | any polycom gurus here? |
11:41.27 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-129-096.dsl.sil.at) |
11:42.01 | feeds | can asterisk convert sound from .wav? I mean automated messages like demo-echotest |
11:53.17 | hi365 | yes, if its the correct wav |
11:57.07 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
12:00.53 | *** join/#asterisk ecm (n=ae@86.121.135.149) |
12:08.13 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
12:15.15 | *** join/#asterisk Segnale007 (n=Pietro@host216-252-dynamic.35-79-r.retail.telecomitalia.it) |
12:58.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
13:12.39 | *** join/#asterisk synchris (n=synchris@athedsl-156469.home.otenet.gr) |
13:17.54 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
13:39.25 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-086-042.dsl.sil.at) |
13:40.50 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
13:40.59 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-246-60.lns10.mel6.internode.on.net) |
13:43.14 | hi365 | wonders if he can set a dhcp server to do inform ONLY (i.e. no ip address) |
13:44.04 | mankash | what is sip show registry |
14:11.45 | *** join/#asterisk ecm (n=ae@86.121.135.33) |
14:11.54 | ecm | Hello, all. |
14:13.54 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
14:18.27 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
14:26.33 | *** join/#asterisk af_ (n=getsmart@88-149-230-152.dynamic.ngi.it) |
14:26.43 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-130-180.dsl.sil.at) |
14:28.19 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
14:31.30 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
14:32.36 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
14:32.36 | *** mode/#asterisk [+o russellb] by ChanServ |
14:41.35 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
14:45.34 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
15:06.24 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
15:10.06 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
15:13.34 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
15:19.41 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
15:27.37 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-130-180.dsl.sil.at) |
15:31.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
15:36.34 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-83-rrdg-esr-2.dynamic.isadsl.co.za) |
15:36.39 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-199-83-rrdg-esr-2.dynamic.isadsl.co.za) |
15:38.33 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
15:38.49 | jaytee | morning russell |
15:39.01 | jaytee | mornin [TK]D-Fender |
15:39.16 | [TK]D-Fender | *yawn |
15:39.58 | jaytee | don't do that! I just everything on the breakfast buffet menu but I can't nap or I'll miss my flight. |
15:40.01 | riddlebox | morning guys |
15:40.17 | riddlebox | jaytee, where you at? |
15:41.03 | jaytee | Huntsville |
15:44.02 | *** join/#asterisk ThOr101 (n=bthorson@pool-96-255-184-162.washdc.fios.verizon.net) |
15:44.45 | ThOr101 | Can 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. |
15:52.18 | jaytee | don'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. |
15:54.14 | ThOr101 | fair enough. Thanks. |
15:55.34 | jaytee | compiling 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. |
15:56.14 | ThOr101 | Oh, 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 :-) |
15:56.17 | jaytee | there's one for CentOS on the WIKI which is very much like Fedora since they're both built based on Red Hat |
15:57.26 | ThOr101 | I 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. |
15:58.58 | hi365 | anyone good with linux dhcp? |
15:59.10 | hi365 | im wondering: can I add option to be distributed to a class, or do they have to be in a pool? |
16:00.39 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591469.dsl.bell.ca) |
16:03.12 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
16:11.53 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
16:12.34 | *** join/#asterisk pluesch0r (n=pluesch0@vie-078-142-131-134.dsl.sil.at) |
16:16.01 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
16:17.21 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
16:17.46 | nextime | is in * 1.4 STREAM FILE agi command stable, or i must use EXEC PLAYBACK instead? |
16:22.42 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
16:24.05 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
16:31.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
16:31.49 | tzafrir_laptop | ThOr101, that is a specific issue of Fedora. This is not the case with Debian and SUSE |
16:32.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:33.13 | ThOr101 | Thanks 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. |
16:33.27 | ThOr101 | I'm in a 5V slot with external power. |
16:33.44 | ThOr101 | I removed all the addon cards. Wierd. I wonder if the card got friend somehow. |
16:34.21 | tzafrir_laptop | IIRC that module won't use a card without modules unless explicitly told so |
16:34.40 | ThOr101 | Yeah, I put them in my zaptel.conf file. |
16:34.45 | tzafrir_laptop | with: timingonly=1 |
16:34.48 | ThOr101 | That bit me last time I was doing this. |
16:34.52 | ThOr101 | ohh, that's new. |
16:38.20 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
16:39.04 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
16:46.47 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
16:52.30 | *** join/#asterisk telecos (n=sergio@84.166.219.87.dynamic.jazztel.es) |
16:57.03 | *** join/#asterisk KP7 (n=chatzill@dsl001-145-071.phl1.dsl.speakeasy.net) |
17:04.02 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
17:15.38 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-109-092.dsl.sil.at) |
17:18.16 | *** join/#asterisk andresmujica (n=andresmu@190.25.104.186) |
17:22.02 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
17:24.41 | riksta | is murf here? |
17:25.14 | *** join/#asterisk hi365_m (n=hi365@213.151.57.132) |
17:28.30 | *** join/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
17:29.41 | *** join/#asterisk protocols (n=protocol@ip-88-152-40-90.unitymediagroup.de) |
17:32.28 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7) |
17:32.54 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
17:33.43 | *** join/#asterisk ManxPower (n=manxpowe@55.sub-75-202-121.myvzw.com) |
17:37.30 | *** join/#asterisk sah-work (n=Bawbatos@commodity-noc-201.sc08.org) |
17:38.14 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
17:39.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
17:41.41 | lesouvage | Has anyone tried this phone http://www.globalsourcesdirect.com/servlet/the-1977/VoIP-Phone-For-SIP-fdsh-IAX2/Detail |
17:42.11 | *** part/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
17:44.59 | root52 | lesouvage: No but the price looks right ;-) |
17:46.00 | WimpMan | Hmm. Seen a big bunch of them on ebay for 19.99. |
17:46.20 | lesouvage | root52: 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 |
17:47.24 | root52 | hahahaha interesting how i do not see a brand name. |
17:48.18 | lesouvage | root52: I think the idea is to brand them yourself. |
17:48.26 | *** join/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
17:48.59 | lesouvage | root52: like "take root52, the different way to go" |
17:49.14 | [TK]D-Fender | Thats an ATCOM phone... PA1688 chipset flaming piece of shit :) |
17:49.23 | ManxPower | Those 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. |
17:49.47 | ManxPower | That's it, the PA168 chipset. |
17:49.58 | [TK]D-Fender | 1688 <- the 88 wasn't a typo :0 |
17:50.27 | ManxPower | [TK]D-Fender: Odd, I thought it was PA168. |
17:50.58 | ManxPower | I bought one one time. |
17:51.06 | [TK]D-Fender | its a crap phone... plasitcy low feature pile of scarp |
17:51.12 | [TK]D-Fender | scrap even |
17:51.19 | ManxPower | Of all the phones I've owned that is the only one that I actually threw into the trash. |
17:51.28 | *** part/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
17:51.28 | *** join/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
17:53.54 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
17:55.18 | lesouvage | ManxPower and [TK]D-Fender: thanks for sharing your experiences. Ordering a full container doesn't sound like a good idea ;-) |
17:55.37 | *** join/#asterisk jaytee (i=cea6ce22@gateway/web/ajax/mibbit.com/x-e7bfc52817099050) |
18:01.36 | [TK]D-Fender | jaytee: Quick flight... |
18:10.27 | jaytee | I'm in the smoking lounge at Huntsville waiting for my flight |
18:11.16 | [TK]D-Fender | jaytee: 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-Fender | jaytee: in the office now to continue on my ^%#$ing budget |
18:11.36 | [TK]D-Fender | heh |
18:11.39 | *** join/#asterisk Carlos_Tico (n=ircap@c-98-200-244-36.hsd1.tx.comcast.net) |
18:12.20 | Carlos_Tico | Does anybody knows if this a problem with pap2 configuration or asterisk? |
18:12.20 | Carlos_Tico | I 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. |
18:13.29 | jaytee | Carlos_Tico: if the time interval for losing the reg varies widely it sounds like a flaky NIC in the pap2 |
18:13.56 | Carlos_Tico | tried with two differents ATA |
18:13.57 | *** join/#asterisk JymmmEMC (n=Jymmm@unaffiliated/jymmm) |
18:14.01 | Carlos_Tico | one pap2 and one spa3102 |
18:14.20 | Carlos_Tico | sometimes happens after the fist registration ends thats it |
18:14.28 | jaytee | Carlos_Tico: but you could try setting the registration interval to a low value so it refreshes more frequently |
18:14.28 | Carlos_Tico | well most of the time |
18:14.37 | Carlos_Tico | nothing happen |
18:14.40 | Carlos_Tico | the same |
18:14.51 | Carlos_Tico | and in the ata debug says AUTH failed |
18:15.29 | JymmmEMC | Dont 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. |
18:16.18 | Carlos_Tico | [jaytee] |
18:16.23 | Carlos_Tico | only happens on the remote ata.... |
18:16.29 | Carlos_Tico | the network ata works perfect |
18:16.50 | jaytee | Carlos, if you're behind a NAT you need to set qualify=yes and canreinvite=no |
18:17.51 | jaytee | Carlos_Tico: that's assuming your pap2 is on the other side of a NAT |
18:18.50 | hi365_m | how 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? |
18:18.59 | Carlos_Tico | let me see |
18:19.10 | Carlos_Tico | yes its a remote ata |
18:19.46 | Carlos_Tico | yes i have those parameters set as qualify and no |
18:21.00 | jaytee | is the network link through an ISP or a private circuit? |
18:21.21 | Carlos_Tico | its linked via ISP |
18:23.23 | [TK]D-Fender | JymmmEMC: How do you want to receive this DID? |
18:24.09 | JymmmEMC | [TK]D-Fender: TCP/IP (sorry, I dont exactly understand the question) |
18:24.52 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
18:24.57 | [TK]D-Fender | JymmmEMC: Ok, from a VoIP provider. Here : |
18:24.59 | [TK]D-Fender | ~itsp |
18:24.59 | jbot | [~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 |
18:25.03 | jbot | [~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 |
18:25.04 | jbot | [~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 |
18:25.45 | Carlos_Tico | [jaytee] and when i try to set up a host=myname.ath.cx ====> works well |
18:25.57 | Carlos_Tico | but when that address changes ip asterisk cant resolve it |
18:26.25 | JymmmEMC | [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-Fender | JymmmEMC: To use Skype? No, Skype is its own little world... |
18:27.16 | JymmmEMC | [TK]D-Fender: =) I just need a PSTN number to do some remote control is all. |
18:27.32 | jaytee | Carlos_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-Fender | JymmmEMC: Ok, but this leaves Skype completely out of the picture, just FYI |
18:28.04 | *** join/#asterisk Unlockgod (n=bob@195.149.30.143) |
18:28.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
18:28.14 | Unlockgod | Hey, wondering if someone could help me with a nat priblem |
18:28.39 | JymmmEMC | [TK]D-Fender: i understand, google mentioned a couple of interfaces is all (http://www.google.com/search?q=asterisk+skype) |
18:28.40 | drmessano | unlockgod? |
18:28.49 | Unlockgod | hi |
18:29.01 | drmessano | What did you unlock? |
18:29.04 | [TK]D-Fender | ~skype |
18:29.05 | jbot | [~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. |
18:29.07 | drmessano | and how was it godlike? |
18:29.07 | [TK]D-Fender | ^^^ |
18:29.29 | Unlockgod | mobile phones (cell phones) |
18:29.31 | [TK]D-Fender | reboxes his Ark Of The Covenant |
18:29.34 | Unlockgod | just a name used for years |
18:29.38 | JymmmEMC | [TK]D-Fender: cool. lol |
18:29.54 | Unlockgod | :p |
18:29.57 | Carlos_Tico | [jaytee] what do you mean wan getting hammered ? |
18:30.07 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
18:30.19 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-37-217.w86-215.abo.wanadoo.fr) |
18:30.35 | JymmmEMC | ~SkypeForAsterisk |
18:30.36 | jbot | [~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-Fender | JymmmEMC: Its locked off, and "pay-per-channel" |
18:31.25 | JymmmEMC | ew |
18:31.26 | [TK]D-Fender | JymmmEMC: Not available just yet |
18:31.43 | [TK]D-Fender | JymmmEMC: You can basically ditch Skype with SIP you know... |
18:31.53 | jaytee | Carlos_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-Fender | JymmmEMC: Skype is for little kiddies and the otherwise VoIP-ignorant |
18:32.25 | JymmmEMC | [TK]D-Fender: to me, it's $60/yr which is fine. |
18:32.38 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
18:32.42 | [TK]D-Fender | JymmmEMC: $60 gets you what exactly? |
18:33.40 | JymmmEMC | [TK]D-Fender: a pstn number and unlimited inbound calling. |
18:34.03 | [TK]D-Fender | JymmmEMC: If you already have this, what do you want * for? |
18:34.21 | tzafrir_laptop | JymmmEMC, take a look at ipkall.com |
18:35.02 | *** join/#asterisk black187 (n=black187@89-212-174-185.dynamic.dsl.t-2.net) |
18:35.02 | Carlos_Tico | [jaytee] is there a way that asterisk can resolve if a dyndyns change the ip ..... |
18:35.15 | black187 | Hello, anybody here with any experience with TLS setup on Asterisk?!? |
18:35.38 | JymmmEMC | [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. |
18:35.50 | drmessano | black187: you got TCP working? |
18:36.04 | [TK]D-Fender | JymmmEMC: Well that sounds like something to do... sure. |
18:36.12 | black187 | nope, still stuck with SSL cert error on Asterisk :( |
18:36.20 | *** part/#asterisk Unlockgod (n=bob@195.149.30.143) |
18:36.23 | black187 | didn't even try the TCP connection |
18:36.27 | Carlos_Tico | aytee ¦ 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 |
18:36.39 | Carlos_Tico | and a router of course..... |
18:37.52 | JymmmEMC | tzafrir_laptop: thank you! |
18:38.55 | JymmmEMC | Does asterisk have a voice response /DTFM decoder feature? |
18:39.01 | [TK]D-Fender | JymmmEMC: Yup |
18:39.16 | [TK]D-Fender | JymmmEMC: IVR's are basic stuff |
18:39.19 | black187 | Did 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-Fender | JymmmEMC: Give the book a good read |
18:39.28 | [TK]D-Fender | ~book |
18:39.28 | jbot | methinks 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 | ^^^ |
18:40.22 | JymmmEMC | [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-Fender | JymmmEMC: Sure |
18:41.07 | JymmmEMC | hawt damn! =) I prefer hw over sw solutions =) |
18:41.32 | [TK]D-Fender | JymmmEMC: its still hardware... jsut cause its on flash and not a spinning HD doesn't change that fact :) |
18:41.47 | [TK]D-Fender | JymmmEMC: Now if you SOLDERED it.. THEN it would be a "hardware" solution ;) |
18:41.54 | JymmmEMC | [TK]D-Fender: No, no, I *LIKE* hardware solutions |
18:42.03 | [TK]D-Fender | still sofware* |
18:42.18 | [TK]D-Fender | JymmmEMC: Just nit-picking on you.. no worries.. |
18:42.24 | JymmmEMC | Ok, non-moving hardware... how's that =) |
18:42.30 | [TK]D-Fender | JymmmEMC: For what I'm sure your limited needs are it should do OK |
18:42.41 | [TK]D-Fender | JymmmEMC: Yeah that pretty much says it |
18:43.05 | [TK]D-Fender | JymmmEMC: faster-booting, non-spinning, cheap all-in-one box solution. |
18:43.33 | JymmmEMC | you wouldn't know if the DTMF supports: 09*#ABCD by chance would ya? |
18:44.49 | [TK]D-Fender | JymmmEMC: It does |
18:44.56 | [TK]D-Fender | JymmmEMC: Full set |
18:45.19 | JymmmEMC | oh awesome! |
18:45.32 | [TK]D-Fender | JymmmEMC: 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-Fender | JymmmEMC: 4x3 is assured |
18:46.44 | JymmmEMC | [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. |
18:48.07 | JymmmEMC | (like autopatch, if you know what that is) |
18:48.58 | [TK]D-Fender | JymmmEMC: Digits are digits, more you can support the more meaningful & simpler overall entry can be. |
18:50.35 | JymmmEMC | [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. |
18:51.50 | jaytee | Carlos_Tico: sorry, I got into a conversation here at the airport lounge with a NASA engineer. |
18:52.15 | Carlos_Tico | its ok |
18:52.33 | Carlos_Tico | [jaytee] is there a way that asterisk can resolve if a dyndyns change the ip ..... |
18:52.33 | JymmmEMC | [TK]D-Fender: Thanks for the help/info, much appreciated! |
18:52.43 | [TK]D-Fender | JymmmEMC: np |
18:53.23 | jaytee | Carlos_Tico: not that I'm aware of as far as a built in function in * |
18:53.59 | jaytee | i've got to logoff because the crappy battery in this Dell is about to die |
18:54.08 | jaytee | later all |
18:54.14 | Carlos_Tico | ok thanks for all your help |
18:54.52 | [TK]D-Fender | Carlos_Tico: * behind NAT needing to know if IP changed? |
18:55.19 | Carlos_Tico | yeah |
18:55.38 | Carlos_Tico | my remote ata its with host=myname.ath.cx |
18:55.53 | Carlos_Tico | but when it changes asterisk cannot resolve it ... |
18:55.56 | [TK]D-Fender | Carlos_Tico: You should setup your DynDNS client, and then do "externhost=my.dyndns.org" and "externrefresh=60" (seconds) |
18:56.07 | [TK]D-Fender | Carlos_Tico: You need the "externrefresh" |
18:56.16 | [TK]D-Fender | Carlos_Tico: that sets how often * rechecks it |
18:56.26 | Carlos_Tico | where to put that ... ? |
18:56.38 | [TK]D-Fender | Carlos_Tico: right beloh your externhost entry |
18:56.41 | [TK]D-Fender | below* |
18:56.52 | Carlos_Tico | in the trunk config |
18:56.55 | [TK]D-Fender | (under [general] of course) |
18:56.56 | Carlos_Tico | right |
18:56.58 | [TK]D-Fender | Carlos_TRUNK? |
18:57.16 | Carlos_Tico | or in the sip_nat.conf |
18:57.20 | [TK]D-Fender | Carlos_Tico: this is a [general] setting. *'s situation is not trunk-dependant |
18:57.32 | [TK]D-Fender | Carlos_Tico:sip_nat.conf |
18:57.36 | Carlos_Tico | ok |
18:57.41 | [TK]D-Fender | Carlos_Which of course tells me you're using FreePBX... |
18:58.00 | Carlos_Tico | :) |
18:58.40 | [TK]D-Fender | Carlos_Tico: But thats where it goes. You should ahve "canreinvite=no", "nat=yes", "externhost=yourhosthere", "externrefresh=60" |
19:01.13 | Carlos_Tico | ok |
19:01.15 | Carlos_Tico | let me try |
19:02.20 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
19:02.26 | Carlos_Tico | but 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-Fender | Carlos_Tico: Your ATA has its own dyndns? |
19:04.37 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
19:04.49 | Carlos_Tico | yes |
19:04.52 | [TK]D-Fender | Carlos_Tico: the ATA should be registering to YOU and you should be setting its register interval. |
19:04.53 | jaytee | \o/ found an A/C outlet in the smoking lounge to charge the battery |
19:05.11 | [TK]D-Fender | Carlos_Tico: You should never be specifying a host for them. always "host=dynamic" |
19:05.24 | [TK]D-Fender | Carlos_Tico: Go into your ATA and chage the register interval. |
19:05.33 | hi365_m | how much of the polycom serer-side call features (cfw, dnd, etc) are suppported by asterisk? |
19:05.37 | Carlos_Tico | i did but i lost the registration with that |
19:05.45 | [TK]D-Fender | Carlos_Tico: How? |
19:06.01 | *** join/#asterisk JackEStorm (n=no@ip24-252-118-155.no.no.cox.net) |
19:06.55 | Carlos_Tico | keeps loosing the registration |
19:07.12 | Carlos_Tico | so thats why in host i changed to dyndns |
19:07.33 | [TK]D-Fender | Carlos_Tico: that makes no sense. a remote ATA should never have to care where ti is. |
19:08.11 | [TK]D-Fender | Carlos_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. |
19:08.46 | Carlos_Tico | in the syslog serv of the ata says Auth Failed ..... |
19:09.01 | drmessano | ATA can be behind any nat |
19:09.01 | [TK]D-Fender | Carlos_Tico: that should have nothing to do with ANY IP. |
19:09.07 | drmessano | any IP |
19:09.15 | [TK]D-Fender | Carlos_Tico: Auth failed = you set the auth up wrong. |
19:09.15 | drmessano | Thats why it REGISTERS |
19:10.05 | Carlos_Tico | but everyting is fine... i just double checked.... |
19:10.17 | drmessano | Asterisk says its not |
19:10.19 | drmessano | I believe it |
19:10.20 | Carlos_Tico | i reset it registers then after a minut |
19:10.31 | Carlos_Tico | its not register anymore .... |
19:11.08 | drmessano | qualify is off for the extension |
19:11.32 | Carlos_Tico | qualify is yes |
19:11.48 | [TK]D-Fender | Carlos_Tico: Your IP can't be changing every MINUTE. |
19:12.05 | drmessano | Factory reset the ATA |
19:12.07 | drmessano | Then reconfig |
19:12.08 | [TK]D-Fender | Carlos_Tico: You have really misinterpreted your problem. |
19:12.08 | Carlos_Tico | no the ip its not changing at all |
19:12.12 | drmessano | With 3 options |
19:12.17 | drmessano | username, pass, proxy |
19:12.18 | Carlos_Tico | i did.... |
19:12.20 | Carlos_Tico | factory reset |
19:12.21 | Carlos_Tico | pass |
19:12.26 | [TK]D-Fender | Carlos_Tico: then how can DyDNS have anything to do with fixing your probelm? |
19:12.27 | Carlos_Tico | proxy ... and the same... |
19:12.34 | [TK]D-Fender | Carlos_Why were you lokoing at it in the first place?" |
19:13.35 | Carlos_Tico | just to register a remote ata with * |
19:14.00 | [TK]D-Fender | Carlos_Tico: again, your ATA does not need this |
19:14.35 | [TK]D-Fender | Carlos_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) |
19:15.19 | Carlos_Tico | ok wait |
19:18.38 | *** join/#asterisk AndyML (n=quassel@pool-96-227-91-204.phlapa.fios.verizon.net) |
19:18.50 | Carlos_Tico | here you are |
19:18.51 | AndyML | morning drmessano, [TK]D-Fender |
19:18.56 | AndyML | morning Carlos_PHX |
19:18.57 | AndyML | err |
19:19.01 | AndyML | Carlos_Tico: |
19:19.01 | Carlos_Tico | hi :) |
19:19.11 | Carlos_Tico | [[TK]D-Fender] |
19:19.21 | AndyML | guys - I've been working with Carlos so he's asked me to come over here and help him explain his problem. |
19:19.22 | Carlos_Tico | [AndyML] can explain better the situation.... |
19:20.05 | *** join/#asterisk zydoon (n=zydoon@41.225.159.36) |
19:20.14 | drmessano | morning |
19:20.27 | drmessano | Well |
19:20.29 | *** part/#asterisk zydoon (n=zydoon@41.225.159.36) |
19:20.31 | drmessano | If I had to take a guess |
19:20.36 | AndyML | This 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. |
19:20.40 | drmessano | The far end firewall is a POS for SIP |
19:20.50 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
19:20.52 | AndyML | far end being where the ATA is? |
19:20.55 | drmessano | yes |
19:21.01 | AndyML | it IS the ata... |
19:21.05 | drmessano | ... |
19:21.07 | AndyML | which doesn't make you wrong |
19:21.14 | drmessano | he said hes using a PAP2? |
19:21.14 | AndyML | but its ironic at least |
19:22.20 | AndyML | Carlos_Tico: is that ATA the router for that network? |
19:22.20 | Carlos_Tico | no |
19:22.20 | AndyML | Interesting. |
19:22.20 | Carlos_Tico | only the ata part |
19:22.20 | drmessano | So what is the router? |
19:22.20 | [TK]D-Fender | ... |
19:22.27 | AndyML | before we get too deep - just one more piece of explaination |
19:22.27 | Carlos_Tico | the router is a WRT300N Linksys with DDWRT sp1 |
19:22.28 | [TK]D-Fender | Swow us the FAILURE. Too much talk, not enough show. |
19:22.48 | drmessano | Are you on Cable or DSL? |
19:22.52 | AndyML | Carlos_Tico: dig up the screenshot of that auth failed from the ATA's logserver |
19:22.57 | AndyML | DSL |
19:23.04 | Carlos_Tico | DSL is the remote ATA ..... |
19:23.09 | [TK]D-Fender | ?!?! |
19:23.21 | drmessano | Is the router doing the PPPoE? |
19:23.28 | drmessano | The WRT300N? |
19:23.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
19:24.05 | Carlos_Tico | yes |
19:24.08 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
19:24.11 | Carlos_Tico | the router is doing PPPoE |
19:24.12 | drmessano | ok |
19:24.12 | Carlos_Tico | http://picpaste.com/Andy.jpg |
19:24.29 | AndyML | (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...) |
19:25.08 | drmessano | Will other devices work? |
19:25.13 | drmessano | Other locations? |
19:25.56 | Carlos_Tico | yes the x-lite works perfect |
19:26.13 | drmessano | behind the same firewall? |
19:26.17 | Carlos_Tico | yes |
19:26.18 | drmessano | same extension? |
19:26.38 | Carlos_Tico | behind the same firewall other extension ata is 4000 x-lite is 3000 |
19:26.46 | [TK]D-Fender | AndyML: 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 |
19:27.08 | drmessano | Carlos_Tico: Blow up the 4000 entry and recreate it |
19:27.35 | [TK]D-Fender | And yes, take out the 4000 out of the picture and see if things start working. |
19:27.48 | drmessano | Use the ATA on 3000 |
19:27.53 | Carlos_Tico | ok |
19:28.12 | jaytee | still ought to pastebin a sip debug of the fail |
19:28.40 | drmessano | Hes probably got something set wrong in the extension config in FreePBX |
19:28.57 | drmessano | So 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) |
19:29.25 | jaytee | FreePBX? hmmmm |
19:29.32 | drmessano | yes |
19:30.04 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-124-230.dsl.sil.at) |
19:30.49 | AndyML | [TK]D-Fender: will do. |
19:31.21 | drmessano | I went to Goodwill yesterday on the "money" side of town.. |
19:31.23 | jaytee | drmessano: this is interesting but I've gotta get to the gate for boarding time. Let me know later how it turns out, ok? |
19:31.31 | drmessano | Since they have better second hand shit |
19:31.32 | drmessano | Ok, will do |
19:31.38 | jaytee | thanks |
19:32.00 | drmessano | This woman was ARGUING over a pair of stained jeans she wanted to buy for $3 and cut up for something |
19:32.09 | jaytee | drmessano, [TK]D-Fender later guys |
19:32.22 | [TK]D-Fender | jaytee: lter |
19:32.24 | jaytee | Carlos_Tico: good luck |
19:32.25 | drmessano | They offered her 10% off due to the stain, which is all goodwill will back down from stuff.. it's goodwill, afterall |
19:32.41 | drmessano | So she argued, and argued |
19:32.44 | Carlos_Tico | thanks jaytee |
19:32.48 | Carlos_Tico | jaytee |
19:32.49 | drmessano | Then paid for it |
19:32.52 | [TK]D-Fender | drmessano: Goodwill neads a big stick to beat ill-will people with |
19:33.01 | drmessano | Then went outside |
19:33.08 | drmessano | Got into her $35000 SUV |
19:33.12 | drmessano | and stormed out of the parking lot |
19:34.25 | [TK]D-Fender | drmessano: sombitch |
19:34.44 | [TK]D-Fender | drmessano: probably burned as much in gas slamming the pedal in anger leaving. |
19:39.07 | drmessano | Yes |
19:40.03 | drmessano | Honestly, if something clearly overpriced I dont mind telling them |
19:40.12 | drmessano | Better to knock a buck off than to not sell it |
19:40.18 | drmessano | But They gave her the 10% |
19:41.25 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.186) |
19:43.35 | Daejeo | will it be useful for initiating the call (DISA)l via email/SMS on asterisk? |
19:44.21 | [TK]D-Fender | Daejeo: You tell us |
19:44.50 | [TK]D-Fender | Daejeo: I wouldn't consider it useful if I didn't have access to SMS / e-mail |
19:45.07 | Daejeo | it is very useful |
19:45.13 | drmessano | or if my mail server didnt have thumbs |
19:45.25 | Daejeo | would you like to know how? |
19:45.35 | Daejeo | it is not useful americans |
19:45.39 | Daejeo | for* |
19:46.23 | drmessano | Running that through my IRC drama filter, I think he just called me useless |
19:46.25 | Daejeo | it is hard to find a DID numbers in south asia. |
19:46.31 | Carlos_Tico | ok here is the ATA configuration .... http://picpaste.com/ATA.jpg |
19:46.33 | drmessano | spits in a bucker |
19:46.41 | drmessano | Carlos_Tico: Did you put it on 3000? |
19:46.46 | Carlos_Tico | yes |
19:46.48 | Carlos_Tico | now is 3000 |
19:46.52 | drmessano | Did it work? |
19:47.19 | Daejeo | but most of the people have a cellphone phone |
19:47.45 | drmessano | Ug |
19:47.51 | drmessano | Use auth ID NO |
19:47.52 | Daejeo | so sms can be used instead of DID number |
19:47.58 | drmessano | Get rid of the auth stuff |
19:48.10 | Daejeo | [TK]D-Fender: any comments? |
19:48.16 | drmessano | Use auth ID NO and get rid of the auth username |
19:48.17 | drmessano | and it will work |
19:48.21 | drmessano | Youre not using AUTH |
19:48.54 | drmessano | It registers, then fails to auth |
19:49.08 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
19:49.15 | drmessano | Problem solved |
19:49.58 | AndyML | for 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... |
19:50.15 | Carlos_Tico | we used yesterday with Andy |
19:50.31 | Carlos_Tico | and before we were not using .... |
19:50.49 | Carlos_Tico | so it was only for testing purpouses |
19:50.58 | drmessano | Authentication is autheticating.. not just registering |
19:51.08 | drmessano | Authentication is more a security mechanism |
19:51.55 | AndyML | how do you require it or not require it in asterisk? |
19:52.02 | AndyML | auth=md5 ? |
19:52.08 | drmessano | yes |
19:52.13 | Carlos_Tico | i already tried with that no luck |
19:52.14 | drmessano | authusername I believe it is, etc |
19:52.16 | Carlos_Tico | same happens |
19:52.28 | drmessano | Carlos_Tico: Get rid of aith |
19:52.30 | drmessano | auth |
19:52.33 | Carlos_Tico | ok |
19:52.34 | drmessano | Dont try to make it work, dump it |
19:52.52 | drmessano | You need THREEE |
19:52.53 | drmessano | 3 |
19:52.56 | drmessano | TRES |
19:53.03 | drmessano | 3 params in the ATA, as I have told you for days |
19:53.11 | drmessano | Username, password, proxy |
19:53.30 | AndyML | thats it |
19:53.33 | AndyML | just those three |
19:53.34 | drmessano | Yes |
19:53.34 | Carlos_Tico | it was like that ... we changed only for testing purposes |
19:53.47 | drmessano | anything else is tweaking |
19:53.54 | drmessano | Some meaningful, some not |
19:53.57 | drmessano | Codec, etc |
19:54.08 | drmessano | But 3 parms will get an ATA regged |
19:54.22 | drmessano | 0 vs 1, work vs not work |
19:54.50 | AndyML | drmessano: 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? |
19:55.14 | drmessano | good god |
19:55.42 | drmessano | if theres that much latency, registration lag is a non-issue.. the device wont be USABLE |
19:57.05 | AndyML | its usually at the lower end, and it works okay for their purposes (keeping family in touch, etc) we just need to keep it registering... |
19:58.20 | root52 | If 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) |
19:59.11 | drmessano | Register expires = 60? |
19:59.17 | drmessano | Who set that? |
20:00.12 | Carlos_Tico | it was just for testing |
20:00.22 | Carlos_Tico | the original value was 600 |
20:00.45 | drmessano | when I say things like "Factory reset it and put this in" |
20:00.50 | drmessano | I am not kidding |
20:01.59 | drmessano | What else is set for "testing"? |
20:02.06 | Carlos_Tico | nothing else |
20:02.16 | drmessano | How do I know that isn't another lie? |
20:02.27 | drmessano | You seem to like to not do things or say you've done things.. |
20:02.42 | drmessano | It's impossible to work off known constants |
20:03.13 | drmessano | I know these devices pretty well, as do others.. and by not setting things to known states, they cannot be troubleshot effectively |
20:03.19 | drmessano | I guess you dont understand that |
20:03.26 | Carlos_Tico | sorry ... those were move only for testing |
20:05.35 | AndyML | drmessano: I haven't worked with this particular device much and made some changes... |
20:05.44 | AndyML | register expires = 60 was me to get your sip debug |
20:05.51 | AndyML | err [TK]D-Fender's sip debug |
20:05.55 | AndyML | http://www.pastebin.ca/1258023 |
20:06.44 | AndyML | Carlos_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-Fender | AndyML: wheres ther egister attempt? |
20:06.54 | drmessano | I 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) |
20:07.18 | Carlos_Tico | i can call them to reset to factory defaults |
20:07.25 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
20:07.34 | linagee | wtf? i was using polycom 2.2.0 and it took like 5+ minutes to load sip.ld! |
20:07.46 | linagee | now i upgraded to 3.1.1 and it takes under 30 seconds. huh?? |
20:07.47 | AndyML | [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-Fender | AndyML: 1-psten? this was 400 a few minutes ago? WTF? |
20:08.10 | [TK]D-Fender | 4000* |
20:08.19 | drmessano | Thats the PSTN FXO |
20:08.21 | AndyML | its is an FXO/FXS device. neither side is workign |
20:08.22 | drmessano | Theres an FXS |
20:08.30 | drmessano | Forget the FXO |
20:08.39 | [TK]D-Fender | Ok, I've hit my limit of dealing with bait&switch debugging. |
20:08.40 | AndyML | they're both failing the same way at essentially the same time |
20:09.28 | [TK]D-Fender | goes off to do something productive. |
20:09.39 | drmessano | Reset the boxes, set the extensions in FreePBX up again |
20:09.51 | drmessano | Put in proxy, user, pass |
20:09.53 | drmessano | and then test |
20:09.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:09.58 | AndyML | will do. |
20:10.05 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
20:10.17 | drmessano | There's too much tweaking going on here |
20:10.21 | drmessano | Need constants |
20:10.39 | AndyML | [TK]D-Fender: sorry for the headache I guess... |
20:10.41 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
20:13.48 | drmessano | AndyML: 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) |
20:17.26 | AndyML | I'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) |
20:18.02 | AndyML | and it didn't show the registrations properly, so i asked how to enable that level of debug, but got no answer... |
20:18.28 | Kdas | i 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-Fender | Kdas: NAT involved somewhere along the way? |
20:19.04 | [TK]D-Fender | ~sipnat |
20:19.05 | jbot | [~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 | ^^^^^ |
20:19.33 | Kdas | [TK]D-Fender, yes its behind nat |
20:19.40 | [TK]D-Fender | Kdas: read up |
20:19.56 | Kdas | [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) |
20:38.02 | *** join/#asterisk sah-work (n=Bawbatos@commodity-noc-201.sc08.org) |
20:43.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:44.03 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
20:50.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:52.07 | *** join/#asterisk andresmujica (n=andresmu@190.25.104.186) |
20:53.59 | *** join/#asterisk korihor (n=korihor@200-71-162-1.genericrev.telcel.net.ve) |
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) |
21:25.59 | root52 | Hi, 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. |
21:26.25 | kimo_sabe | hmm, g729 outgoign calls are dropping on me when the person picked up. What am I missing? |
21:27.06 | kimo_sabe | I bought a license, and it works for inbound calls |
21:27.39 | [TK]D-Fender | root52: Depends if * is accepting the call at all or not. Enable SIP debug and see |
21:28.42 | root52 | Ahhh good call |
21:30.19 | kimo_sabe | the asterisk console isn't giving me any hints as to why it's dropping the call |
21:30.52 | kimo_sabe | * is SIP Declining the call |
21:31.18 | *** join/#asterisk arpu (n=arpu@chello084114022060.14.vie.surfer.at) |
21:32.14 | [TK]D-Fender | kimo_sabe: Go looka t your call at verbose 10, sip debug enabled. |
21:32.31 | kimo_sabe | [TK]D-Fender: ah |
21:34.10 | [TK]D-Fender | It was a sad day when the Lone Ranger realized that "kimosabe" translated as "horse's ass" :p |
21:35.52 | kimo_sabe | [TK]D-Fender: indeed :) |
21:37.33 | kimo_sabe | hmm, still just "Zap/1 answer SIP/blah" |
21:37.40 | kimo_sabe | hangup Zap/1 |
21:39.46 | [TK]D-Fender | kimo_sabe: Still you not showing us the call in a pastebin with the kind of debug you were suggested to do. |
21:40.16 | kimo_sabe | [TK]D-Fender: oh, you wanted it in pastebin, one sec |
21:40.30 | [TK]D-Fender | kimo_sabe: Are you expecting help when you show nothing? |
21:41.03 | kimo_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-Fender | kimo_sabe: I did. Told you verbsoe 10 & sip debug. |
21:44.43 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
21:46.13 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-086-012.dsl.sil.at) |
21:54.02 | kimo_sabe | http://pastebin.ca/1258225 |
21:57.24 | kimo_sabe | um 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) |
22:07.15 | kimo_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-Fender | kimo_sabe: You say you bought a license. Now show us that it is installed and available |
22:07.33 | kimo_sabe | ppbx*CLI> show g729 |
22:07.33 | kimo_sabe | 0/0 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) |
22:09.17 | kimo_sabe | bug 8781 is looking similar |
22:09.21 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
22:11.52 | kimo_sabe | hmm, 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-Fender | kimo_sabe: Trya a call from 307 to something else that clearly requires tranascoding. |
22:21.23 | baliktad | Can someone have a look and tell me if this call is failing because of me or my provider? http://pastebin.ca/1258285 |
22:24.50 | protocols | is there any good howto for nvfaxdetect? |
22:24.50 | [TK]D-Fender | baliktad: SIP/2.0 483 Too Many Hops <- badsetup on your end |
22:25.03 | kimo_sabe | [TK]D-Fender: hmm, neat. I can call the fax machine and listen to it squeal |
22:25.27 | baliktad | all I can see is the one hop from me to my provider. Why are they claiming too many hops? |
22:25.56 | [TK]D-Fender | baliktad: PASTEBIN YOUR PEER ENTRY |
22:25.59 | [TK]D-Fender | darn caps |
22:26.52 | baliktad | http://pastebin.ca/1258296 |
22:28.44 | [TK]D-Fender | baliktad: yOU'RE BEHIND nat, AREN'T YOU? |
22:28.59 | baliktad | I am |
22:29.10 | [TK]D-Fender | baliktad: You should have put "nat=no" for this entry then. |
22:30.36 | baliktad | same result (yes, I reloaded sip module) |
22:33.51 | [TK]D-Fender | baliktad: Now PB your sip.conf masking only passwords |
22:36.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:38.47 | kimo_sabe | what could be different between Zap/1 & Zap/28 that would stop g729 transcoding from working? |
22:41.29 | baliktad | ok, once again, I'm an idiot |
22:42.00 | baliktad | I had a duplicate account in my sip.conf from an earlier copy/paste |
22:42.01 | kimo_sabe | and Zap/1 -> SIP/307 does work |
22:43.12 | *** join/#asterisk jov4n (n=jovan@87.19.163.231) |
22:43.14 | jov4n | Hi |
22:49.35 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
22:51.05 | drmessano | Anyone know is a 16vac power supply for a 480e is a "standard" of some sort of Aastra |
22:51.10 | drmessano | if* |
22:52.05 | drmessano | I have been trying to find one online.. thinking maybe all the 480s or maybe ALL their phones use the same supply |
22:53.47 | drmessano | Ah, think I found one |
22:53.50 | drmessano | $20 bleh |
22:54.15 | drmessano | Anyone need an Aastra 480e? lol |
23:06.09 | [TK]D-Fender | kimo_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) |
23:18.10 | kimo_sabe | [TK]D-Fender: this is the only phone with allow=g729, and there are no other active channels |
23:18.55 | [TK]D-Fender | kimo_sabe: I suggst you do a channel dump prior, and post and test again |
23:19.16 | kimo_sabe | [TK]D-Fender: channel dump? |
23:19.30 | [TK]D-Fender | show channels concise |
23:20.38 | kimo_sabe | nothing before or after, this while ringing: Zap/1-1!from-zaptel!8913738!1!Dialing!AppDial!(Outgoing Line)!8913738!!3!0!(None) |
23:20.41 | kimo_sabe | SIP/307-b8778d90!macro-dialout-trunk!s!25!Ring!Dial!ZAP/g0/5208913738|300|W!5208886740!!3!0!(None) |
23:21.31 | [TK]D-Fender | kimo_sabe: ok, I'm at a bit of a loss right now... |
23:22.51 | interfaithquest | Fender, i can see your hard core with asterisk ! great |
23:23.20 | interfaithquest | so i heard the audio on twit.tv of mark spencer.. cool |
23:24.41 | interfaithquest | anyway.. are you using dundi ? i am wondering how to random users can find each other , call from * to * using iax for example |
23:25.17 | interfaithquest | dundi seems more of a gateway sharing system, than anything else ? |
23:27.05 | kimo_sabe | ok, 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-Fender | kimo_sabe: Go ask them why it doesn't transcode then :) |
23:33.12 | kimo_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 |
23:33.37 | kimo_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-Fender | kimo_sabe: You saying that the OSS one works where your paid one doesn't? |
23:33.47 | kimo_sabe | [TK]D-Fender: yup |
23:33.54 | [TK]D-Fender | kimo_sabe: You should call Digium about that.. |
23:35.04 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
23:35.33 | kimo_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-Fender | kimo_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-Fender | interfaithquest: DUNDi ? meh... |
23:48.45 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
23:48.49 | interfaithquest | some registry of asterisk servers ? for * to * calling |
23:50.01 | drmessano | Dundi is a good idea for that |
23:50.06 | interfaithquest | i guess one could simply click on a webpage for a click to call service |
23:50.16 | drmessano | Intercompany |
23:50.22 | drmessano | or load balancing in a way |
23:51.00 | interfaithquest | anyway.. why call unless you find the party of interest |
23:51.42 | interfaithquest | someway to post your * for voip connections |
23:52.11 | drmessano | You can post your URI |
23:53.08 | interfaithquest | yes , others have tried sip proxies.. where they give YOU ..there ID.. ha ha..who is in the middle.. of the universe ? |
23:54.20 | drmessano | Im talking direct SIP dialing |
23:54.23 | interfaithquest | so 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 |
23:54.25 | drmessano | To your PBX |
23:54.55 | interfaithquest | sip is crippled by the NAT issues world wide |
23:55.01 | drmessano | heh |
23:55.05 | drmessano | SIP is just fine |
23:55.20 | drmessano | "NAT Issues" fall into the realm of improper setup |
23:55.32 | interfaithquest | tell that to the milliions of skype user |
23:55.47 | drmessano | What does Skype have to do with SIP? |
23:55.50 | drmessano | or NAT? |
23:56.10 | interfaithquest | if 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 |
23:56.22 | interfaithquest | instead 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) |
23:56.39 | drmessano | Skype users dont have SIP connectivity due to the closed nature of Skype |
23:56.47 | drmessano | Has nothing to do with SIP |
23:57.14 | interfaithquest | so where is the global universal protocol, the whole world can use for FOSSL voip ? |
23:57.49 | drmessano | Dunno, are you gonna invent it? |
23:57.55 | drmessano | Lemme know how it goes |
23:58.00 | interfaithquest | will do |
23:58.10 | interfaithquest | it's time has come for sure |
23:58.39 | drmessano | Im not sure how to break this to you |
23:58.49 | drmessano | But SIP URI dialing is well on its way to becoming a standard |
23:59.05 | interfaithquest | well google seems to be happy pushing XMPP jingle for its part |
23:59.14 | interfaithquest | yes.. sip may break thru with ICE |
23:59.20 | TrentCreek | Okay..I got the trunk working. I used the example context provided by les.net |
23:59.30 | drmessano | Ever heard of a little company called Microsoft who has been pushing SIP dialing? |
23:59.51 | interfaithquest | true billy goat does have desire to rule the world |