00:05.48 | bmoraca_work | why do telephone companies suck so much to deal with? uhg |
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00:11.16 | WIMPy | I've got an isse with HWEC "wct4xxp 0000:03:08.0: VPM450: firmware dahdi-fw-oct6114-064.bin not available from userspace". What might be missing? More info at http://wimpy.yeti.dk/pastebin |
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00:17.06 | raden | bmoraca_work, u tell me |
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00:29.20 | *** join/#asterisk ldiamond (46331fee@gateway/web/freenode/ip.70.51.31.238) |
00:29.56 | ldiamond | One of my device (3G phone) keeps failing to receive incoming calls due to the following: [Dec 22 19:28:27] NOTICE[26982]: chan_sip.c:18223 handle_response_peerpoke: Peer '1' is now Lagged. (2770ms / 2000ms) |
00:30.07 | ldiamond | Anyone have an idea how to solve such an issue? |
00:30.52 | WIMPy | Get a better connection (or allow a bigger lag). |
00:31.08 | ldiamond | Where can we allow bigger lag? |
00:31.20 | ldiamond | as for the connection... well it's 3G... what can I do? |
00:35.59 | WIMPy | qualify=<ms> |
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00:38.05 | devmod | Uhm asterisk is running but it suddenly stopped listening on port 5060. Any idea hwo to debug that issue? |
00:38.45 | WIMPy | devmod: Read that before, but it hasn't happened to me. |
00:39.08 | devmod | It's really weird. hmm |
00:39.45 | WIMPy | Ugh. Dahdi sending shit again :-( |
00:41.12 | wizard171 | WIMPy: my apologies, got sidetracked, and my dahdi_(fu) just ain't found it ... even after looking over what you posted ... however, if its doing that while its running, maybe you have a timing issue? (ie: slips?) what does it log when it happens? |
00:42.04 | WIMPy | wizard171: No, it only dies so when running dahdi_cfg. Thereafter everything is fine. |
00:42.11 | WIMPy | s/dies/does/ |
00:42.50 | WIMPy | Voice quality is flawless. |
00:43.36 | wizard171 | no ticking, popping, or sharp snapping noises of any kind? |
00:44.47 | WIMPy | nope |
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00:45.27 | wizard171 | devmod: ah, does "module show like sip" still show ... something ? |
00:49.39 | devmod | wizard171, I killed it :/ I had forgotten to enable any kind of logging |
00:53.51 | wizard171 | devmod: that ... would make debugging ... hard ... ;) |
00:54.49 | wizard171 | WIMPy: I am wading thru "userspace" ... if I go to sleep, poke me ... :) |
00:54.54 | devmod | wizard171, right, I wasnt expecting sip to just die :P |
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00:55.27 | wizard171 | devmod: :) ... no one ever does ... until it does ... eh? |
00:55.33 | WIMPy | wizard171: I wouldn't. Sleep ist important! |
00:55.42 | devmod | haha it just crashed again |
00:55.51 | devmod | i mean , sip crashed |
00:56.27 | devmod | chan_sip.so Session Initiation Protocol (SIP) 2 |
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01:01.24 | wizard171 | WIMPy: did you build the kernel yourself? (ie: did you "enable userspace") |
01:02.12 | WIMPy | wizard171: I did. What option/feature is that? |
01:02.48 | WIMPy | Iguessed thare's something missing, I just couldn't find out, what. |
01:05.07 | wizard171 | WIMPy: in udev or hotplug ... I am looking for exactly where ... which kernel are you using ? |
01:05.26 | WIMPy | I don't have udev running. |
01:05.35 | WIMPy | 2.6.36.2 |
01:06.27 | wizard171 | WIMPy: ah, unless I am mistaken, udev is required for hotplug, and loading of firmware from "userspace" ... can you start it ? |
01:06.41 | WIMPy | OH. |
01:09.20 | WIMPy | wct4xxp 0000:03:08.0: VPM450: Present and operational servicing 2 span(s) |
01:09.59 | wizard171 | Yeah! Congrats! |
01:10.07 | WIMPy | I thought that were two differen things, as no hardware detection is taking place. |
01:10.19 | WIMPy | Thank you very much. |
01:10.45 | wizard171 | Sorry for the delay ... got sidetracked on my end ... glad its working ! |
01:11.17 | WIMPy | That's fast enough for me :-) |
01:13.58 | WIMPy | Now I can try to find ouy why it is sending a redirecting number. |
01:15.33 | WIMPy | BTW: Is tehre any reason, why it defaults to |
01:15.36 | WIMPy | wct4xxp 0000:03:08.0: VPM450: hardware DTMF disabled. |
01:25.17 | wizard171 | ah, cause asterisk has enough problems with DTMF in the first place ... :) no, really, most only turn it on if asterisk gives DTMF recognition problems ... I forget the setting to turn it on tho ... :) |
01:25.48 | WIMPy | modinfo helps :-) |
01:26.01 | WIMPy | I just wondered if there was a reason. |
01:27.10 | wizard171 | None I have come across ... other than the "asterisk is supposed to do it" kind of thinking that was built into it in the beginning ... |
01:27.43 | WIMPy | Sounds like I want it on then. |
01:28.35 | wizard171 | ah, only if you have issues "recognizing" dtmf ... then I would give it a try ... otherwise, it just might ... *give* ... you problems with DTMF ... eh ? :) |
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01:29.37 | orioni | hi there, i want to analyse the acd/asr for the cdr on mysql |
01:29.52 | orioni | is there any ready script/application i can try ? |
01:29.55 | WIMPy | You think the algorithm running on the DSP is not as good as that in Asterisk? |
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01:32.22 | wizard171 | What, who me (as he looks around suspiciously) ... *good* ... is entirely ... subjective ... ;) nor, would I *ever* suggest such about the "left" hand at asterisk and the "right" hand at digium ... eh? |
01:33.57 | wizard171 | Ergo, for me, its about ... "does it work = no" then " try letting the DSP do it instead" ... as far as I have gone on that road ... :) |
01:34.50 | WIMPy | I'll try it out. |
01:35.47 | wizard171 | Well ... I hope ... all your buttons get pressed ... just the way you want ... eh ? :) |
01:35.59 | orioni | is there anyone that does reporting on the cdr`s on mysql for acd/asr ? |
01:36.26 | orioni | i would like some example since my queries are executed after 11 seconds and i want sth faster than that |
01:36.40 | WIMPy | If it can do THAT ... |
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02:34.05 | dandate2 | how do you do a conference call with an ATA device? i.e. sip 1 is talking to sip2, sip1 adds sip3 to the conversation |
02:34.22 | dandate2 | its so easy with xlite =@ |
02:35.17 | WIMPy | Transfer bot to a MeetMe() or ConfBridge() extension. |
02:35.22 | WIMPy | both |
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02:38.02 | dandate2 | this is soemthing i have to configure at the * end or the ATA end? |
02:38.11 | dandate2 | i would have assumed the ATA end since xlite has it built in |
02:40.40 | dandate2 | this wouldnt be practical for us since typically sip2 is on a cell phone and mentally challenged |
02:40.52 | dandate2 | when using * conference, each person has to dial a code huh |
02:43.06 | dandate2 | so it is the ata device that does the 3 way calling, god damn this thing for not having a manual |
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02:51.15 | dandate2 | nm i guess we just had to use the flash button heh |
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02:52.42 | matt_vu | looks around |
02:55.23 | wizard171 | matt_vu: ... you are ... it ... |
02:56.28 | matt_vu | asterisk is a pbx? |
02:56.45 | wizard171 | only if you configure it that way ... ;) |
02:57.27 | wizard171 | I have no luck ... configuring it to bring me women ... :) |
02:57.44 | matt_vu | i see, so all the discussions here are about 'as a SIP'? |
02:58.01 | matt_vu | :)@wizard... try harder |
02:59.36 | wizard171 | ah, SIP ... the coffee, other times we IAX2, skinny, gtalk, h323, etc. ... some even ... dahdi ... |
03:01.26 | matt_vu | do u mean gtalk is on asterisk? |
03:03.31 | wizard171 | yes, you can gtalk on asterisk ... |
03:04.03 | matt_vu | ic |
03:05.56 | wizard171 | I guess you could say ... you can ... jabber while you jingle on your gtalk ... ;) |
03:06.10 | matt_vu | :) |
03:15.55 | ldiamond | Is there a setting to have sip poke retries when theres no response? |
03:17.16 | ldiamond | I'm talking about the peer poke here |
03:18.27 | wizard171 | Ah, if you set "qualify=xxx" where xxx = some numeric value, that is how often it "pokes" ... is that where you are going ? |
03:19.14 | wizard171 | If you have it set "qualify=yes" ... you get 2000ms (I think, as default) ... |
03:19.17 | WIMPy | qualify is the maximum time to wait for an answer. |
03:19.25 | WIMPy | yes |
03:19.51 | cmendes0101 | Is there a way to pass variable using originate command in CLI? |
03:20.11 | wizard171 | It's "how often to check" in "peer" terms, I believe ... |
03:20.31 | WIMPy | No, that's qualifyfreq. |
03:21.17 | WIMPy | And I think thare is a qualifyfailfreq or something as well. Don't see that ATM. |
03:22.48 | WIMPy | Hmm. Maybe that was IAX only. |
03:23.40 | wizard171 | You are right "qualify=xxx" time it has to respond inside of (so to speak) "qualifyfreq=" how often to "poke" the other side ... I don't see a "fail" ... |
03:24.09 | WIMPy | No. Looks like that is indeed iax only. |
03:26.03 | wizard171 | cmendes0101: what do you have in mind? |
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03:26.39 | wizard171 | s/mind?/mind? (source vs. value)/ |
03:27.16 | WIMPy | But sip does have different values internally as well. I just can't find a way to configure it. |
03:29.38 | wizard171 | well ... the source code, usually does not start ... telling stories ... until after you compile it and run it ... eh ? :) |
03:30.05 | cmendes0101 | wizard171: I have this "originate local/s@join application playback FILE", join context makes it join a meetme conference. I would like to pass a variable so it knows what conference number to go into. Or is there another way to have that application enter the meetme conference? |
03:30.26 | ldiamond | Looking at this: https://issues.asterisk.org/bug_view_page.php?bug_id=16936 |
03:30.26 | WIMPy | s/re it/re it at runtime/ |
03:30.40 | ldiamond | Maybe this might help fix my issue |
03:30.50 | ldiamond | upgrading to newer version. |
03:30.56 | cmnky | Millions affected as Skype goes down |
03:30.56 | cmnky | Wed Dec 22, 6:04 pm ET |
03:33.38 | wizard171 | cmendes0101: that originate is something a human is typing ? (if not, can the script substitute what you want?) |
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03:35.17 | wizard171 | ldiamond: what version are you running? |
03:35.23 | cmendes0101 | wizard171: well it is a script but I'm not sure how else to pass the application(audio) into the meetme conference. This works except it is only going into the hardcoded conference number in the context |
03:35.45 | ldiamond | 1.6.2-5 or something |
03:35.57 | ldiamond | I am updating it now. |
03:41.39 | wizard171 | cmendes0101: ah, instead of "s", can the script specify "this1" or "that1", which can be defined as "exten => this1,blah" , etc. ? |
03:42.31 | wizard171 | it would mean many different versions ... or, creative dialplan ... :) |
03:42.49 | ldiamond | 1.6.2.5-0ubuntu1.1 |
03:44.16 | wizard171 | ldiamond: the "fix" did not make it until 1.6.2.8 (according to my changelog) ... |
03:45.23 | wizard171 | or should I have said the "fix of the fix, which was fixed ...", etc. |
03:45.48 | cmendes0101 | wizard171: Haha that might be something I can work out as a solution. I can always try. I'm guessing passing a variable through there isnt really clear cut or possible |
03:46.47 | ldiamond | arghhh, and ubuntu doesnt update fast enough |
03:46.58 | wizard171 | cmendes0101: well ... "its a command line" ... it can "vary" by whatever your script feeds it ... :) |
03:47.37 | wizard171 | ldiamond: ah, but ubuntu does ... much else ... very well ... eh? |
03:48.17 | ldiamond | wizard171: yea, it's fine, but I'm wondering if I should go through the trouble of building from source and fitting this into my current setup |
03:49.28 | wizard171 | ldiamond: (did I just have "dejavu"? ala "The Matrix") ... Goto the Source my friend ... it only starts ... telling stories, after you compile it ... :) |
03:49.43 | cmendes0101 | wizard171: actually if I do something like exten => _XXXX, 1, Meetme(${exten}) that should work right? |
03:49.48 | cmendes0101 | well I guess ill just try it lol |
03:50.05 | p3nguin | pbx_config won't like spaces. |
03:50.51 | wizard171 | cmendes0101: yeah, and you should "filter" that "${exten}" for "valid" as well ... ;) |
03:51.22 | ldiamond | wizard171: I got FreePBX setup as well, don't want everything to start failing and spend 12hrs fixing it (like last time :S) |
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03:54.28 | wizard171 | ldiamond: shouldn't be a problem ... "./configure --prefix=/usr" (I think) ... for a "point" release, you just need to "compile" on top of FreePBX's version ... |
03:55.08 | wizard171 | ldiamond: just don't "make samples" ... eh? |
03:55.38 | ldiamond | I'm scared about sudo apt-get remove asterisk.... my config will most likely be gone.. |
03:57.24 | p3nguin | lrn2cp or lrn2rsync |
03:57.39 | wizard171 | ldiamond: "host# cp -r /etc/asterisk /path/to/keepit" ... same/same for "dahdi", etc. |
03:58.06 | ldiamond | yea, it's the 'etc' part there that scares me :p |
03:58.58 | wizard171 | ldiamond: (sigh) ... yeah, that's the one I usually forget ... :) |
03:59.29 | wizard171 | lf you have another machine to practice with ... that would be good ... |
03:59.34 | p3nguin | Really, I wouldn't expect apt-get to delete your files unless you use purge. |
03:59.50 | p3nguin | I'd still save my backups, though. |
04:00.00 | p3nguin | It only takes a second to make a backup. |
04:00.05 | p3nguin | literally |
04:00.22 | WIMPy | wats that machine! |
04:00.40 | wizard171 | or, you could "image" the drive onto another ... |
04:01.07 | WIMPy | Maybe use RAID? *cough* |
04:01.52 | p3nguin | If you delete a file on RAID, it gets deleted everywhere. |
04:02.13 | wizard171 | just much faster ... than mirror ... or JBOD ... |
04:02.58 | p3nguin | It takes only a fraction of a second to clear an inode from being used. |
04:04.54 | wizard171 | ah, but, compiling from the source = much better ... IMHO ... |
04:05.13 | p3nguin | The packages were built using the same source. |
04:06.07 | wizard171 | meaning, worth the effort, to get what you want/need ... and sometimes, optimized for what you have ... eh ? |
04:07.27 | p3nguin | Unless you're running 25-year-old hardware, you're not likely to see any performance gains by "optimizing" your own compile. |
04:07.53 | p3nguin | The packages provide the exact same software that you'll get if you compile it yourself. |
04:08.31 | WIMPy | Either the package won't even run un 15 year old hardware or you have a good chance of about 10%. |
04:09.53 | wizard171 | I am not knocking "packages" ... I am just acknowledging that the "compile" process has been designed to take advantage of the current machines CPU/Compiler, etc. In most cases it will NOT be "the same" as the package ... |
04:11.28 | p3nguin | And my point was that if you are using hardware made in this century, you're not going to gain anything by optimizing on your own personal hardware. |
04:11.50 | p3nguin | If you think I'm wrong, benchmark it and show me the results. |
04:14.08 | wizard171 | hmm ... I have more faith in ... "make the compile work" ... than "what someone else compiled" ... all day long, every day, no benchmark needed ... :) (no offense meant) I just like (and understand) the end result ... better ... IMHO ... |
04:18.13 | p3nguin | I'd have a fair amount of faith in the guys who maintain the packages every day/week/month/year for a given distro. |
04:19.32 | p3nguin | Not that it really makes any difference for me, since I build my own packages, but I think I made my point. |
04:20.14 | wizard171 | well ...the compile does take into account what is there on the machine, which, in some cases "makes it work" ... better ... than the decisions that were made in the compile for the "package" ... |
04:26.18 | wizard171 | or, perhaps, I should say "the contents of your machine" may be "different" than the machine you "compile and run" it on ... and sometimes that produces a ... better ... end result ... again, no offense, I will admit it ... I like to "compile" it ... :) |
04:27.15 | wizard171 | makes me feel ... special ... ;) |
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04:36.46 | brainiac | Does anyone know why I can dial out on TE110P, but can't receive incoming calls? Do I have to reboot the server after I add the DID |
04:36.49 | brainiac | ? |
04:39.41 | WIMPy | "dialplan reload" |
04:42.51 | wizard171 | brainiac: does "add the DID" mean changes to "chan_dahdi.conf"? (very little in there can be changed without a "restart" at least) ... |
04:44.32 | WIMPy | Yes, took me quite some time yesterday to remember that restarting Asterisk isn't enough when reconfiguring interfaces. |
04:45.17 | p3nguin | You ought to be able to reload the necessary module(s) and get changes to take effect, I'd think. |
04:45.41 | brainiac | I've been restarting this thing with every little change. I'm getting this on the CDR: -- Extension '2100' in context 'TE110P/4' from '6022495750' does not exist. Rejecting call on channel 0/1, span 1 |
04:45.56 | p3nguin | I don't know; I just don't reboot my computers. It irritates me, for some reason. |
04:46.03 | WIMPy | Yes, I had to rmmod and modprobe. But I'm not used to that. |
04:46.08 | wizard171 | And you thought all that ... "modprobe"(ing) ... was just for ... fun ... |
04:47.58 | wizard171 | brainiac: I am pretty sure you do not have a "context" [TE110P/4] in your "dialplan"? What "context" does "2100" exist in? (and perhaps where should you be sending "inbound" calls into?) |
04:48.53 | brainiac | inbound calls are going to a particular extension depending on the number dialed. |
04:53.24 | wizard171 | well, okay then, in your "chan_dahdi.conf" in the [channels] section, the part that applies to "port 4" ... what does the text just after "context=???" say ? (and does that value exist in your "dialplan", in a way that "makes calls go to a particular extension, depending on the number dialed") (no offense meant, just trying to sort it out) |
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04:58.16 | brainiac | context=TE110P/4 (zapata.conf not chan_dahdi.conf) |
04:59.39 | brainiac | This is only a 1 channel card (TE110P). I think this should say context=TE110P/1. Right? |
05:00.08 | WIMPy | You can call it whatever you like. |
05:00.13 | wizard171 | brainiac: my bad, should have thought of "zapata" ... as for the "TE110P/4" part ... that should probably be ... something else ... what "context" does your "2100" live in? |
05:00.40 | wizard171 | in "extensions.conf" or "extensions.ael" ? |
05:01.32 | brainiac | it lives in globals |
05:01.40 | brainiac | extensions.conf |
05:01.41 | p3nguin | Fail. |
05:03.25 | wizard171 | Okay ... in "extensions.conf" ... you REALLY need some other "[context-name]" style entries ... or an "#include ..." that has them ? |
05:03.55 | brainiac | ok |
05:04.11 | p3nguin | You need [general], [globals], and THEN contexts where you put useful extensions. |
05:04.31 | wizard171 | where the "[context-name]" part has some "exten => ... " type entries ... underneath it ... |
05:04.39 | p3nguin | I'd even add a [default] in there after [globals] just to be thorough. |
05:07.32 | wizard171 | Once you define a "[context-of-your-choice]" type entry, with "exten => ... " type entries ... change the "zapata.conf" to have the same "[context-of-your-choice]" without the "[]" ... and see if that works out ... |
05:07.44 | brainiac | ok |
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05:07.50 | p3nguin | ~book |
05:07.50 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
05:07.54 | p3nguin | I guess no one reads anymore. |
05:08.16 | p3nguin | Too much work, perhaps. |
05:08.29 | azlon | where can i find an asterisk ubuntu 10.10 repo? |
05:09.36 | drmessano | Download and install from source.. you don't want a repo |
05:09.44 | azlon | hrmm |
05:09.46 | azlon | ok |
05:09.54 | azlon | i will search for a guide to install from source |
05:09.55 | azlon | thanks |
05:15.05 | brainiac | thanks for the help everyone. BTW, I am reading an Asterisk book called "Practical Asterisk 1.4 & 1.6 From Beginner to Expert" by Stephan Wintermeyer and Stephen Bosch. Great Book. |
05:15.28 | azlon | hehehehehe |
05:15.35 | azlon | nae! |
05:15.39 | azlon | now click on living room |
05:15.43 | WIMPy | Too many books. |
05:15.49 | azlon | i set ioops |
05:15.52 | azlon | wrong room |
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05:18.12 | wizard171 | Yeah, I gave up "reading" and started "experimenting" ... and there are those (unkind souls) that say, my "experiments" never should have made it into "production" ... :) |
05:18.59 | wizard171 | Just what I get for creating ... "art" ... eh? |
05:19.52 | wizard171 | or, maybe they think I left the "f" out of that word ... ? (who knows?) |
05:22.38 | WIMPy | That's what I heard until I remembered to reload the kernel drivers. |
05:23.14 | wizard171 | Yeah ... those kernels ... will drive you, alright ... LOL |
05:24.16 | wizard171 | I am SURE ... "popcorn" ... had nothing to do with it ... eh? |
05:25.14 | WIMPy | It wasn't me. It was my dahdi. |
05:25.37 | wizard171 | I know ... it just wants to ... zapata ... ? |
05:28.19 | wizard171 | I am going to ... go sleep ... before this devolves into ... "extensions" ... ;) I have enjoyed, until we meet again ... enjoy! |
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06:52.43 | drmessano | O.o |
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07:27.37 | schmidts | good morning |
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07:32.28 | drmessano | gm |
07:33.40 | schmidts | everybody allready in christmas mood? |
07:47.36 | *** join/#asterisk Russ (~russ@ip68-111-71-150.oc.oc.cox.net) |
07:50.47 | Russ | anyone deal with support issues with Level 3? |
07:55.12 | drmessano | schmidts, I hate christmas, so no |
07:58.03 | schmidts | drmessano i can understand this :D |
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08:56.20 | arekm | hi, I'm in "s" context. Can I somehow figure out what extension was dialled anyway? |
08:57.47 | schmidts | arekm you mean the s extension not context, but to answer the question not really cause the EXTEN variable will be set to s. you have to create another matching extension like _X. then you will see in EXTEN what has been dialed |
08:59.07 | *** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au) |
08:59.27 | Corydon76-home | Unless he's using a Macro, in which case, the EXTEN will be in ${MACRO_EXTEN} |
09:11.00 | joobie | hey gusy |
09:11.48 | *** join/#asterisk devyll (~paul@thpallady.net.hostway.ro) |
09:14.01 | devyll | hello all. Does anyone know how can I use originate command with the application Goto ? I'm trying originate dahdi/g1/number1 application Goto arg1,arg2,arg3 but it behaves really weird .. anyone tried this combination ? My objective is to originate a call to a number, and go to a specific point (extension) from the dialplan and also at the same time pass an argument (destination number) .. doing that wilth "extension" instead of "application" wo |
09:14.02 | devyll | n't let me pass the argument .. trying with "application" Goto .. it behaves really weird . |
09:21.20 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-41-188.lsanca.fios.verizon.net) |
09:30.39 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.182.59) |
09:39.15 | kaldemar | devyll: why not use context, extension and priority in the originate? |
09:40.03 | kaldemar | and use an extension that lets you have a part of the extension as the argument. |
10:03.53 | devyll | kaldemar, yeah .. it would work.. resolved with call files though |
10:03.54 | devyll | thanks |
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10:52.27 | *** join/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
10:53.32 | X-Raimo | question about realtime. How asterisk searches for number: local and then realtime or vice versa? |
10:53.53 | schmidts | X-raimo with number you mean extension? |
10:54.03 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
10:57.00 | X-Raimo | schmidts: yes |
10:57.59 | schmidts | x-raimo i dont know if asterisk do a live lookup at each call in realtime or just load it for example at dialplan reload. if its just loaded then it makes no difference |
11:01.52 | X-Raimo | schmidts: we have problem with echo test using srtp. ext. 1999 (echo) is localy defines in extensions.conf. And we dial from 1538 which located in DB. Echo test fails with 488 error. |
11:02.39 | X-Raimo | should we define srtpcapable=yes and encryption=yes for 1999 in sip.conf? |
11:05.41 | schmidts | X-Raimo you are mixing extensions and sip peers/users |
11:06.13 | schmidts | or do you have a peer named 1999 which is a physical phone or server which do the echo test? |
11:06.45 | X-Raimo | schmidts: server's echo test |
11:08.07 | schmidts | X-raimo is it exten => 1999,1,Echo() or exten => 1999,1,Dial(SIP/serverforecho) ? |
11:09.06 | X-Raimo | schmidts: exten => 1999,1,Echo() |
11:09.24 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
11:09.25 | schmidts | ok, sorry but i just want to be sure ;) |
11:10.39 | schmidts | it could be that echo doesnt work with srtp but i am not sure, never used encryption |
11:11.11 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
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11:17.59 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
11:21.43 | X-Raimo | Is phone connected via SRTP uses same codecs as phone connected without SRTP? |
11:22.36 | b14ck | X-Raimo: yes. |
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11:25.12 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
11:25.57 | marksaitis | hey. does anyone know which package /usr/local/ssl came from in centos5? |
11:38.15 | Tim_Toady | marksaitis try yum whatrovides |
11:39.12 | Tim_Toady | but since its in local it must be from some compiled by hand source package |
11:48.21 | *** join/#asterisk coppice (~chatzilla@210.17.219.137) |
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12:10.52 | *** join/#asterisk azlon (~ryan@94.29.137.166) |
12:11.02 | azlon | is there a gui for the new asterisk 1.8? |
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12:16.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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12:45.36 | X-Raimo | Is option encryption in sip.conf has only 2 values: "yes" and "no"? What other options does it have? |
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13:02.12 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
13:02.48 | *** part/#asterisk azlon (~ryan@94.29.137.166) |
13:04.08 | anonymouz666 | 1.4.38 is the latest bug fix release, right? 1.4.39 and so on will be released only if there's security fix |
13:04.47 | leifmadsen | anonymouz666: no, 1.4.39-rc1 is out and will become the next release if there is not an rc2 |
13:04.54 | leifmadsen | if there was a security release, it would be tagged at 1.4.38.1 |
13:05.16 | leifmadsen | only in the 1.6.x series do we increase the numbers on a security release (since 1.6.2.15.1 looks silly) |
13:05.45 | leifmadsen | a security (or minor) release on the 1.8 series would look a lot like the 1.4 series (which is why we have 1.8.1.1 right now) |
13:06.04 | anonymouz666 | 1.6.2.15.1-rc3 looks fine |
13:06.06 | anonymouz666 | hehe |
13:06.17 | leifmadsen | the 1.4 and 1.6.2 series have had their bug fix statuses extended to April 2011 |
13:06.26 | leifmadsen | we would never have a 1.6.2.15.1-rc3 |
13:06.35 | leifmadsen | the .1 on the end would be released directly without an RCs |
13:06.50 | leifmadsen | because it would be either a bad regression, or a security release which would not warrant a release candidate |
13:06.55 | anonymouz666 | ahh it was extented then, I remeber reading the latest 1.4 release on 21 december 2010. |
13:07.02 | leifmadsen | yes, that was before astricon :) |
13:07.18 | leifmadsen | we extended it at astricon so people would have more time to migrate/test 1.8 |
13:07.45 | leifmadsen | anonymouz666: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
13:08.34 | leifmadsen | ~asterisk-versions |
13:08.34 | infobot | it has been said that asterisk-versions is Information about the maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at http://www.asterisk.org/asterisk-versions |
13:10.02 | leifmadsen | infobot: no, asterisk-versions is <reply> Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
13:10.02 | infobot | okay, leifmadsen |
13:10.09 | leifmadsen | ~asterisk-versions |
13:10.09 | infobot | Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
13:10.37 | leifmadsen | ~coffee |
13:10.37 | infobot | from memory, coffee is the reason the net exists, the drug of choice for a GNU generation, http://www.chez.com/emarsden/downloads/coffee.el, /usr/share/doc/HOWTO/en-html/mini/Coffee.html, geiseri's favorite beverage |
13:10.42 | russellb | leifmadsen: back to work |
13:11.00 | leifmadsen | russellb: I'm tired of writing about queues :) |
13:11.09 | russellb | leifmadsen: write about DUNDi then |
13:11.13 | leifmadsen | bah! |
13:11.22 | leifmadsen | this book should write itself |
13:11.34 | leifmadsen | I just want my shiny nickle! |
13:11.40 | russellb | heh |
13:11.57 | leifmadsen | queues is already up to 32 pages... |
13:11.58 | russellb | a whole nickel?! |
13:12.12 | russellb | I was only promised a pretty penny. |
13:12.16 | leifmadsen | ya, but it'll be a US nickel which is worth less than 5c |
13:12.25 | leifmadsen | russellb: you get upgraded to nickel on the third book |
13:12.54 | russellb | goes back to work :-p |
13:12.56 | schmidts | you guys think you get paid for this? i though you have to pay to be allowed :D |
13:13.07 | russellb | pretty much |
13:13.34 | schmidts | :D |
13:13.34 | leifmadsen | joinempty! ooo! |
13:14.13 | drmessano | Hows that entire chapter on Answering Machine Detection coming along? |
13:14.21 | drmessano | That would make a good Dec 23rd gift |
13:16.31 | *** join/#asterisk Corydon76-home (white@c-69-137-80-31.hsd1.tn.comcast.net) |
13:16.31 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
13:17.51 | coppice | a whole chapter of evil |
13:18.23 | leifmadsen | drmessano: we're not even bothering :) |
13:18.32 | leifmadsen | maybe in the 4th edition... |
13:19.04 | drmessano | or in the red leather bound 666th edition? MUHAHAHAHA |
13:19.43 | file | leifmadsen, less complaining more writing or the electrocution will continue! |
13:19.44 | leifmadsen | yes |
13:19.54 | leifmadsen | file: and you're not kidding |
13:20.20 | coppice | there's nothing wrong with the ethics of AMD that cash can't cure |
13:20.35 | leifmadsen | :) |
13:25.11 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
13:25.38 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
13:28.01 | coppice | does anyone do a usable AMD for *? |
13:28.48 | russellb | not that I know of |
13:29.06 | coppice | surely people like vicidial must have something |
13:29.25 | russellb | true ... I think I just pretend that evil like that doesn't exist |
13:30.19 | anonymouz666 | hah |
13:30.24 | anonymouz666 | poor vicidial |
13:30.33 | *** join/#asterisk wizard171 (~wizard@h47.50.20.98.dynamic.ip.windstream.net) |
13:31.12 | SuPrSluG | I tried using AMD. 80% is about the best you can hope for. |
13:31.32 | russellb | I've heard some optimistic accounts of 90% ... |
13:31.55 | coppice | you mean the app that is in the * distribution? that one is too mickey mouse to be of any real value |
13:31.57 | SuPrSluG | that's pretty good. |
13:32.15 | russellb | Asterisk: The Mickey Mouse PBX |
13:32.26 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn110.78-98-245.t-com.sk) |
13:33.17 | coppice | shhhh. don't let disney hear you say that |
13:33.41 | russellb | eep |
13:40.05 | russellb | Mickey Mouse is a registered trademark of Disney. Asterisk is not endorsed by Disney (though they probably use it). (Please don't sue me.) |
13:40.31 | wizard171 | Nah, that would be ... Goofy ... :) |
13:40.51 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn111.78-98-213.t-com.sk) |
13:41.22 | path | is it possible to detect when someone hangs up? |
13:41.38 | path | I need to use GotoIf and some Channel status |
13:42.03 | path | i.e. dial some extension if someone hangs up |
13:47.54 | *** join/#asterisk grkblood13 (967da669@gateway/web/freenode/ip.150.125.166.105) |
13:48.35 | grkblood13 | i need suggestions on a softphone that supports conferencing and has the ability to adjust the volume of the peopel in the conference individually, anyone know a softphone that can do that? |
13:49.15 | marksaitis | in sip.conf, how do I configure a user to have a password? i.e. user [101] |
13:49.54 | SuPrSluG | secret= |
13:49.59 | wizard171 | marksaitis: its a "secret=xxx" ... shh don't tell ... |
13:50.12 | *** join/#asterisk slackytude (~slacky@drms-590df9cf.pool.mediaWays.net) |
13:53.02 | marksaitis | ok |
13:54.39 | schmidts | marksaitis but you should not use users like 101 or something, cause its not secure |
13:56.25 | path | how can I dial to some extension if somebody hangs up before any dtmf? |
13:58.14 | wizard171 | path: sorry, I do not understand ... you mean, once the phone goes "off-hook"? |
13:59.02 | path | the thing is: a user is supposed to answer some questions on a ivr. But what if this user hangs up before any dtmf. |
13:59.33 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
13:59.58 | path | I need to be able to log that anyway, so I want to grab that info |
14:00.39 | marksaitis | how to set CLI verbosity to its max? I get file open fail error, wanna know more info |
14:00.54 | path | core set verbose 50 |
14:00.57 | path | core set debug 50 |
14:01.06 | marksaitis | thanx |
14:01.28 | X-Raimo | is 50 maximum debug value? |
14:01.40 | marksaitis | same error still :))) whats the point in this verbosity thing |
14:03.04 | wizard171 | path: I presume you are using "Dial" ... and either "M(acro)" or "G(osub)" ... and in there, using "Read" to capture what they press? |
14:03.34 | path | yes |
14:03.46 | path | but I forgot cdr exists.. so |
14:04.26 | wizard171 | and perhaps ${DIALSTATUS} |
14:04.30 | path | I tried it |
14:05.04 | path | e.g. exten => s,1,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?s,12) |
14:06.18 | wizard171 | marksaitis: the point to verbosity ... sometimes you get ... more words ... for the same error ... eh? |
14:06.25 | marksaitis | anybody: |
14:06.26 | marksaitis | <PROTECTED> |
14:06.58 | marksaitis | goodness sake, I already gave chmod 777 and chown asterisk to all these cert files, every single one :) |
14:07.03 | marksaitis | still no go |
14:07.10 | X-Raimo | marksaitis: I had this problem yesterday. |
14:07.18 | marksaitis | what did u do? |
14:07.19 | marksaitis | :) |
14:07.38 | X-Raimo | marksaitis: I solved it by changing my client's softphone |
14:07.54 | X-Raimo | phoner lite works well. |
14:08.00 | marksaitis | so u recon its softphone |
14:08.05 | marksaitis | phoner lite? tls srtp? |
14:08.20 | X-Raimo | marksaitis: yes,sir |
14:08.23 | marksaitis | damn softphones, whats wrong with these f*ckers |
14:08.47 | X-Raimo | portgo gives an error |
14:08.56 | marksaitis | I just tried bria on iphone |
14:09.16 | X-Raimo | now I have problems with srtp |
14:09.17 | marksaitis | why the hell does it not say what file.... |
14:09.18 | marksaitis | ;] |
14:09.31 | X-Raimo | still dunno |
14:10.07 | X-Raimo | there is *-file exists. It's actually an error message |
14:10.37 | marksaitis | I am totally getting pissed of by this tls cosmos |
14:10.39 | marksaitis | ;]]] |
14:10.47 | Katty | morning |
14:14.04 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
14:14.11 | marksaitis | dear oh dear |
14:16.37 | Katty | hello my asterisk does not work at all how to fix pls |
14:17.12 | seanbright | Katty: #asterisk |
14:17.15 | seanbright | oh wait... |
14:17.25 | X-Raimo | Katty: what error does it gives? |
14:17.37 | Katty | X-Raimo: what is error |
14:17.39 | marksaitis | Just tried from other softphone |
14:17.40 | marksaitis | [Dec 23 14:18:42] NOTICE[17742]: chan_sip.c:23459 handle_request_register: Registration from '"redc" <sip:101@redc>' failed for '192.168.3.97:16328' - No matching peer found |
14:17.40 | Katty | X-Raimo: it does not work |
14:18.26 | X-Raimo | marksaitis: transport=tls |
14:18.57 | X-Raimo | Katty: give us /var/log/asterisk/messages |
14:19.00 | marksaitis | X-Raimo ? |
14:19.09 | marksaitis | I have transport=tls |
14:19.10 | Katty | giggles hysterically |
14:19.18 | Katty | aww you thought i was serious |
14:19.21 | Katty | that's /so/ cute |
14:19.39 | X-Raimo | marksaitis: you should enter this in sip.conf |
14:19.48 | Katty | pats X-Raimo |
14:20.04 | X-Raimo | in peer's section |
14:20.46 | marksaitis | http://pastebin.com/PjZDKPei |
14:20.50 | marksaitis | please have a look |
14:24.25 | X-Raimo | marksaitis: there is what I have: http://pastebin.com/RG3jQHGW |
14:24.44 | X-Raimo | maybe it gonna be useful for you |
14:25.13 | marksaitis | ok. thanks for sharing |
14:27.03 | *** join/#asterisk azlon (~ryan@94.29.137.166) |
14:27.15 | azlon | anybody here use PBX in a Flash? |
14:28.11 | luckman212 | azlon: yes I do |
14:28.48 | azlon | luckman212: i just installed PIAF and for some reason i cant log in on the webmin |
14:29.02 | azlon | luckman212: everything i have read says that my user/pass should be the same as my linux login |
14:29.27 | azlon | luckman212: i ran passwd-master so they would all be the same, but it doesnt work... keeps giving me the login prompt over and over |
14:29.54 | azlon | luckman212: am i doing something wrong? |
14:30.04 | luckman212 | have you tried running "passwd-maint" from the SSH console? |
14:30.23 | azlon | no |
14:30.28 | azlon | is that different from passwd-master? |
14:31.06 | azlon | trying it now |
14:31.07 | luckman212 | hmm, well normally you would use the maint pwd to log into the web gui |
14:31.21 | luckman212 | passwd-master should work also |
14:31.35 | luckman212 | which piaf version did you install? |
14:31.40 | azlon | uhmm |
14:31.46 | azlon | the newest one... let me find the version |
14:32.10 | azlon | 17554 |
14:32.19 | *** join/#asterisk Vrtigo1 (~Vrtigo1@vpn.lpga.com) |
14:33.16 | azlon | sweet! |
14:33.18 | azlon | it worked! |
14:33.19 | azlon | thanks! |
14:33.24 | marksaitis | can anybody tell me PLEASE what does this mean: |
14:33.24 | marksaitis | [Dec 23 14:34:43] NOTICE[18027]: chan_sip.c:23459 handle_request_register: Registration from '"redc" <sip:101@redc>' failed for '192.168.3.97:16328' - No matching peer found |
14:33.29 | marksaitis | PLEASSSSSSEEEEEEE |
14:33.33 | Vrtigo1 | anyone know why the record app has a 20 minute limit in 1.6.2.2? |
14:33.47 | luckman212 | azlon: cool, glad to hear! |
14:33.50 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
14:34.02 | luckman212 | enjoy your Pbx in a flash... its a lot of fun to play with |
14:34.57 | azlon | luckman212: do you know where i can get info on how to setup a GV account? |
14:35.07 | azlon | yeah, this thing is pretty impressive so far |
14:35.21 | azlon | i am in kuwait right now and want to setup a system so my guys can call back to the states easily |
14:36.17 | luckman212 | GV support is built into the version you have, just run the incredible pbx installer and it walks you thru... |
14:36.33 | marksaitis | I did a shit on this fucking tls |
14:37.08 | azlon | hrmm |
14:37.15 | azlon | ok |
14:37.36 | luckman212 | azlon: http://pastebin.com/kPLPx3S0 |
14:37.50 | azlon | thanks! |
14:39.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
14:40.49 | *** join/#asterisk vasanthv16 (~Nithya@216.195.17.249) |
14:40.52 | *** part/#asterisk vasanthv16 (~Nithya@216.195.17.249) |
14:50.09 | azlon | luckman212: is incredible PBX an addon to PIAF or are they two different things? |
14:50.28 | luckman212 | it's an addon for PIAF |
14:50.33 | azlon | ok |
14:51.13 | luckman212 | gives you a bunch of extra features. read up: http://nerdvittles.com/?p=712 |
14:51.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
14:54.02 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-zbzrrhsxxcuhoolm) |
14:55.53 | frigidzephyr | WIMPy: did you ever resolve teh "not available from userspace" issue with the TE220 firmware? |
14:56.04 | frigidzephyr | if not, you might consider calling Digium technical support to report the issue |
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15:24.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:24.17 | Katty | stretches |
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15:41.07 | *** join/#asterisk jbesclapez (~jbesclape@41.208.164.78) |
15:41.33 | jbesclapez | good morning or evening depending on where you are.... |
15:41.46 | hurdman | how use dialplan function : round and/or floor into an 1.6.* ? i can't found anything |
15:41.48 | jbesclapez | It is my first step in this channel... |
15:42.58 | p3nguin | ~ask |
15:42.58 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
15:43.02 | p3nguin | ~answers |
15:43.02 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
15:43.09 | p3nguin | jbesclapez: ^^^^^^^^^^ |
15:43.16 | Katty | omnomnoms snack |
15:43.43 | jbesclapez | I have a question regarding my new install of Asterisk 1.4.22.1 on my NAS. I am not totaly new to asterisk because i have a working Trixbox. Nevertheless i have a question. |
15:44.16 | p3nguin | If you think you know how to use Asterisk because you've used Trixbox, you may very well be wrong. |
15:44.35 | schmidts | i wish you all merry christmas! |
15:44.50 | jbesclapez | probably p3nguin you are right... i was just pointing that to tell you that i am not a total newbie |
15:45.00 | jbesclapez | schmidts: thanks and to you too. :-) |
15:46.13 | jbesclapez | However, i have softphones receiving OUT and INBOUND calls ok. Just my PAP2 is not running with asterisk. And i cannot find the solutions... it is just receiving call but NO outbounds.... maybe a route issue... no idea. |
15:47.12 | p3nguin | You'll need to show us some useful evidence that supports your claim before we can even try to guess what is wrong. |
15:47.32 | _Corey_ | so, we had a weird one this morning... (I say weird, but I've seen this before elsewhere but only very rarely--but I digress) |
15:47.45 | _Corey_ | we had an Asterisk box go nutty rotating logs |
15:47.47 | _Corey_ | Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
15:48.12 | jbesclapez | p3nguin: what would you like me to proove you exactly? |
15:48.17 | p3nguin | Asterisk doesn't have built-in log rotation, right? |
15:48.24 | _Corey_ | Basically, what happens is /var/log/asterisk fills up with zero-length files |
15:48.42 | p3nguin | jbesclapez: A sip debug would be a great place to start. |
15:48.48 | _Corey_ | p3nguin: nope |
15:48.59 | jbesclapez | OK... |
15:49.08 | _Corey_ | I've seen this a few times before on other systems but have never been able to explain it |
15:49.14 | p3nguin | _corey_: Nope it doesn't, or nope I'm not right? |
15:49.34 | jbesclapez | let me google how to do a sip debug :-) |
15:49.34 | _Corey_ | p3nguin: no log rotation is configured in logger.cnf |
15:49.48 | jbesclapez | you mean the sip.conf? |
15:50.00 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
15:50.29 | _Corey_ | so, when I say /var/log/asterisk fills up, I mean thousands and thousands of files |
15:50.47 | _Corey_ | so, Asterisk becomes unresponsive and needs to be killed hard |
15:51.15 | _Corey_ | <PROTECTED> |
15:51.25 | _Corey_ | and then Asterisk runs happily once again |
15:51.51 | _Corey_ | When I see "SIGXFSZ (Exceeded file size limit)" I think kernel-level ulimit type stuff |
15:52.07 | p3nguin | jbesclapez: I haven't asked for your sip.conf yet. I want to see a sip debug of the failed call from that ATA that can't send calls outbound. |
15:52.11 | _Corey_ | but nothing really jumps out... the original log files weren't too big |
15:52.32 | _Corey_ | anyhow, just curious if anyone else has seen this one |
15:52.50 | p3nguin | jbesclapez: Depending on your asterisk version, you enable sip debug from the CLI by "sip set debug" or "sip set debug on". |
15:52.59 | _Corey_ | seems to happen a couple times a year |
15:53.11 | jbesclapez | how do i send a sip debug? where to look for that info? in the call detail records - this one? |
15:53.43 | p3nguin | And then pastebin the entire debug output after you collect it. |
15:53.43 | leifmadsen | _Corey_: sounds like your logger.conf is setup to log lots of information |
15:54.02 | jbesclapez | i am googling this ... :-) |
15:54.05 | _Corey_ | leifmadsen: yeah: full => notice,warning,error,verbose,debug |
15:54.06 | leifmadsen | _Corey_: I've seen that happen -- it's because you're probably logging a lot of data to that directory. Try rotating the files and such. |
15:54.10 | leifmadsen | well there you go |
15:54.14 | leifmadsen | that's why it fills up |
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15:54.29 | *** part/#asterisk WonTu (~WonTu@p57B53D77.dip.t-dialin.net) |
15:54.51 | _Corey_ | leifmadsen: Well, what's weird is the original full log was a few hundred megs, which isn't too big--I've had MUCH MUCH bigger files |
15:55.16 | _Corey_ | leifmadsen: And when it goes into a tailspin, it's basically rotating zero-length files |
15:55.21 | p3nguin | jbesclapez: There's nothing to google. I gave you exact and precise instructions. |
15:55.27 | leifmadsen | shrugs |
15:55.31 | _Corey_ | yeah |
15:55.34 | *** join/#asterisk makafre (40568d85@gateway/web/freenode/ip.64.86.141.133) |
15:55.39 | _Corey_ | lol |
15:55.47 | _Corey_ | I pretty much had the same reaction |
15:56.03 | jbesclapez | then i must be stupid because i dont know what you mean or where to find the sip debug.... |
15:56.17 | p3nguin | But asterisk doesn't have rotation, so maybe you can stop saying that asterisk is rotating files, because it isn't. |
15:56.34 | _Corey_ | I've probably seen this five or six times over the last couple years on various systems |
15:56.39 | p3nguin | jbesclapez: <p3nguin> jbesclapez: Depending on your asterisk version, you enable sip debug from the CLI by "sip set debug" |
15:56.42 | p3nguin | <PROTECTED> |
15:57.02 | p3nguin | Note: someone who knows how to use asterisk would know this. |
15:57.11 | p3nguin | Just sayin'. |
15:57.15 | _Corey_ | p3nguin: They Asterisk CLI is saying "Asterisk Event Logger restarted" and "Rotated Logs Per SIGXFSZ (Exceeded file size limit)" |
15:57.18 | leifmadsen | or read the documentation :) |
15:57.23 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
15:57.31 | makafre | good morning guys; can someone please remind me how to install the jabberreceive command? e.g. https://reviewboard.asterisk.org/r/88/diff/ |
15:57.45 | _Corey_ | p3nguin: So I'd say it's rotating them, but it's not configured to per-se |
15:57.51 | leifmadsen | makafre: you need lib-iksemel and whatever other deps |
15:57.57 | leifmadsen | makafre: http://ofps.oreilly.com |
15:58.16 | leifmadsen | makafre: don't install via a patch, because it's already part of asterisk |
15:58.18 | p3nguin | I no longer expect people to be how to read documentation. All your hard work of writing docs is in vain. |
15:58.33 | makafre | @leifmadsen: ah, serious, I thought I had to merge it |
15:58.35 | leifmadsen | makafre: notice how that review is closed as "submitted" |
15:58.46 | leifmadsen | that means it's been submitted and merged into the project already |
15:58.49 | leifmadsen | p3nguin: amen |
15:59.14 | makafre | @leifmadsen: I see....and how do I check in which release it was merged initially? |
15:59.22 | p3nguin | s/be how/know how/ |
15:59.34 | p3nguin | When did Asterisk get built-in log rotation? I use 1.4 and I have to let my system rotate the logs. |
16:00.05 | leifmadsen | makafre: click on the link to the bug: https://issues.asterisk.org/view.php?id=12569 |
16:00.23 | _Corey_ | I know the 'logger rotate' command had been around a log time |
16:00.28 | _Corey_ | s/log/long/ |
16:00.29 | leifmadsen | makafre: then scroll near the bottom and look at the message that says what revision it was merged in |
16:00.44 | leifmadsen | September 25, 2009 |
16:01.24 | _Corey_ | This customer's box is actually running 1.2.14 |
16:02.13 | jbesclapez | p3nguin: it is now working... i dont understand... probably the provider had a problem... |
16:03.23 | _Corey_ | I'm not going to invest any more time in this one though, just curious to see who else had seen it happen before |
16:03.29 | _Corey_ | thanks guys |
16:04.04 | p3nguin | jbesclapez: Magic, maybe? Most people believe that's how asterisk works. |
16:04.23 | tvc123 | p3nguin: most people believe that about computers in general |
16:04.29 | p3nguin | s/asterisk works/computers work/ |
16:04.36 | p3nguin | better? |
16:04.45 | jbesclapez | yea because most of the people are stupid right ? ;-) |
16:05.03 | p3nguin | A lot of people are pretty stupid, but I don't know if most people are. |
16:05.14 | tvc123 | jbesclapez: their not stupid until they piss me off ;) |
16:05.49 | makafre | @leifmadsen: thanks , I now see the info, so since I use 1.6.2.15 I should get access to this new function, thats right, thanks; but its being rejected whenever I try to use it from a diaplan, could it jsut be a menuselect issue? |
16:05.54 | jbesclapez | a policeman pisses you off usually... does it mean he is stupid??? (note to myself; bad exemple) |
16:05.59 | _Corey_ | my girlfriend believes that if she speaks soothingly to her pink laptop that it will run better |
16:06.25 | p3nguin | That's kind of retarded... |
16:06.27 | jbesclapez | she is using a mac probably no? |
16:06.34 | leifmadsen | makafre: ya it is included there I'm pretty sure. It's being rejected because you probably haven't met the dependencies, which means it wasn't compiled. Check menuselect to make sure it is selected and that you have installed the dependencies and re-run ./configure |
16:06.35 | Katty | peeks in |
16:06.43 | _Corey_ | no, they're not available in pink from apple i don't think |
16:06.48 | leifmadsen | makafre: again.... ---> http://ofps.oreilly.com |
16:06.52 | _Corey_ | lol |
16:06.53 | p3nguin | It's pink, so everyone should know you have to get rough with it and sometimes smack it around a bit. |
16:06.53 | leifmadsen | it's all documented there |
16:06.56 | jbesclapez | lol |
16:07.12 | _Corey_ | I like to feed people tech psychosis |
16:07.19 | makafre | @leifmadsen: thanks a lot, I am on the right way now |
16:07.28 | _Corey_ | i got my mother a roomba and told her she could control how it moved by clapping |
16:07.32 | leifmadsen | _Corey_: my fiancee just smashes hers and I tell her not to do that |
16:07.33 | _Corey_ | that was HILARIOUS |
16:08.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
16:08.46 | jbesclapez | what is a roomba (sorry) |
16:08.55 | leifmadsen | robot vacuum |
16:08.57 | _Corey_ | the robotic vacuum |
16:09.23 | jbesclapez | eh google told me ;-) |
16:10.11 | leifmadsen | ya, you should have just done that first :) |
16:12.05 | jbesclapez | yep i know... sometime i talk too much to humans ;-) |
16:13.46 | _Corey_ | ahh.. the requests to change holiday hours are coming in already |
16:14.10 | _Corey_ | looks like people are going to stop working this afternoon everywhere |
16:14.23 | jbesclapez | OK everybody.... have a nice christmas and thanks especially to p3nguin |
16:14.42 | jbesclapez | bye |
16:15.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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16:30.46 | Katty | wtb vicks vapor rub |
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16:35.21 | makafre | @leifmadsen: funny enough, I have JabberSend working properly. but JABBER_RECEIVE can't be find; I also checked res_jabber.c and searched through it, but didn't.... I am presently using 1.6.2.15 |
16:35.38 | leifmadsen | it's possible JABBER_RECEIVE is only on 1.8 then |
16:35.46 | leifmadsen | did you look in res/funcs for it? |
16:35.59 | makafre | let me verify |
16:36.38 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
16:36.53 | makafre | yep, it's not there |
16:37.12 | leifmadsen | grep -r "JABBER_RECEIVE" * |
16:37.46 | leifmadsen | could probably look in CHANGES to see if it was added |
16:37.50 | leifmadsen | it should list it if it was |
16:37.52 | makafre | it returns nothing, its definetely not there |
16:38.02 | leifmadsen | then it's a 1.8 only feature |
16:38.15 | makafre | okido |
16:38.57 | makafre | then I am stucked :- ) lol |
16:39.10 | azlon | luckman212: when i dial my GV number, nothing happens... i ran the incredible pbx like you said and entered my user/pass |
16:40.03 | luckman212 | azlon: what do you see in your * console |
16:40.50 | azlon | luckman212: nothing |
16:41.03 | *** join/#asterisk Tim_Toady (~moi@77.49.3.151.dsl.dyn.forthnet.gr) |
16:41.03 | azlon | it is ringing right now and it just says "pbx*CLI>" |
16:41.12 | luckman212 | core set verbose 5 |
16:41.32 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
16:41.40 | luckman212 | type that & try again |
16:41.41 | azlon | call it again now? |
16:41.43 | azlon | ok |
16:41.58 | luckman212 | are you trying inbound --> your pbx or outbound? |
16:42.06 | azlon | inbound |
16:42.30 | luckman212 | have you set up your NAT/firewall properly? |
16:43.14 | azlon | uhmm |
16:43.18 | azlon | probably not... |
16:43.28 | azlon | i dont have a firewall yet |
16:43.58 | azlon | im trying to figure out how to copy from this VM |
16:44.03 | azlon | i need to ssh into it |
16:44.25 | luckman212 | you're running your pbx in a vm? |
16:44.39 | azlon | luckman212: just until i figure out how it works |
16:44.43 | azlon | this is just temporary |
16:44.48 | luckman212 | where is this vm? behind your nat? |
16:44.58 | azlon | yes |
16:46.01 | luckman212 | does the pbx inside the vm have a real lan ip (bridged) or is it double-natted? what kind of vm is it, vmware, proxmox etc? |
16:46.20 | azlon | it is bridged |
16:46.29 | azlon | it is VMWare 7 |
16:46.56 | azlon | ok |
16:47.01 | azlon | i ssh'd into it |
16:47.05 | azlon | this is what it is giving me |
16:47.06 | azlon | http://pastebin.com/ci5LunrC |
16:47.26 | luckman212 | that looks ok, now try to make an inbound call |
16:48.13 | azlon | more of the same thing |
16:48.20 | azlon | it just keeps posting that same message over and over |
16:48.43 | leifmadsen | ya that msg isn't related to GV |
16:48.44 | azlon | let me try calling from a different number |
16:48.53 | luckman212 | your inbound call is not even hitting the pbx |
16:48.58 | luckman212 | you've got a NAT/firewall issue |
16:50.03 | luckman212 | have you followed the instructions? |
16:50.19 | luckman212 | you need to fwd UDP 5222 to your pbx for GV inbound |
16:50.47 | azlon | no, i didnt see any instructions |
16:50.53 | azlon | i just followed the pastebin you gave me |
16:50.59 | azlon | then followed the onscreen instructions from there |
16:51.16 | luckman212 | I pasted in the link to the full instructions. maybe you missed it |
16:51.17 | Nugget | http://bash.org/?35339 |
16:51.21 | luckman212 | please read it |
16:51.41 | azlon | ok, brb |
16:51.49 | *** join/#asterisk brah (~brah@host92.200-82-54.telecom.net.ar) |
16:52.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
16:52.25 | azlon | erm... what am i looking at on there? |
16:52.33 | azlon | this is a chat log |
16:55.13 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
16:58.39 | _Corey_ | There's some good stuff on http://bash.org/?top |
16:59.46 | luckman212 | try here.... http://i.imgur.com/jDhrt.png |
17:01.13 | azlon | cool, still reading through the first one |
17:01.23 | azlon | this thing is freaking crazy... it does EVERYTHING! |
17:03.54 | luckman212 | easy now, don't get ahead of yourself |
17:04.13 | luckman212 | :D |
17:11.47 | Qwell | Please tell me you aren't actually trying to use the PBIAF howto for gV.. |
17:12.46 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
17:26.37 | azlon | hrmm |
17:26.39 | azlon | interesting |
17:26.44 | azlon | i think im burned out for the night |
17:27.03 | azlon | thanks for your help, luckman212. ill pick it up again tomorrow |
17:27.44 | luckman212 | sure... come back & let us know how you get on |
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17:30.56 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
17:33.23 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
17:35.07 | anonymouz666 | shrimp |
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17:41.44 | makafre | @leifmadsen: would you recommend installing the JabberReceive application listed here: https://reviewboard.asterisk.org/r/88/diff/?expand=1 as a complement to my 1.6.2.15 install? I am trying to find an alternative... |
17:42.05 | leifmadsen | No, I never recommend backporting something unless you have the skills to do so |
17:42.14 | makafre | ok |
17:42.24 | makafre | sure |
17:43.51 | Qwell | backports leifmadsen to Asterisk 1.2 |
17:43.57 | leifmadsen | oh god no |
17:43.58 | Qwell | deprecates the backported leifmadsen. |
17:44.01 | Qwell | EOL |
17:44.47 | coppice | leifmadsen: so, in general you wouldn't recommend doing something you can't? |
17:44.58 | leifmadsen | coppice: precisely :) |
17:47.19 | paulc | backs away nervously |
17:48.08 | paulc | Playing around with recording.. MixMonitor seems handy and all, but if I want a stereo file with caller in one channel and agent in another, I have to use Monitor and do some magic afterwards, right? (just looking for confirmation I'm on the right track here) |
17:48.45 | leifmadsen | paulc: ya, and there is a switch to actually mix the channels together after. I've never put them on separate left/right channels, but I think that'd be the approach to do it |
17:51.10 | *** part/#asterisk azlon (~ryan@94.29.137.166) |
17:52.41 | paulc | @leifmadsen: thanks.. figured as much, just needed confirmation - feeling a bit fuzzy this morning.. (and the separate channels thing is nice for separation of caller vs "talent" - needed for "coaching opportunities") ;-) |
17:53.05 | leifmadsen | paulc: ya, Monitor() is good when you want separate files for each channel |
17:53.55 | paulc | and is StopMonitor required? This was a new one on me.. documentation talks about recording via Monitor explicity versus via a feature invoked from features.conf |
17:54.56 | leifmadsen | paulc: then you'd use the 'm' flag and the ${MONITOR_EXEC} channel variable to give the commands to sox or soxmix directly yourself so you can mix the files to the left and right channels yourself |
17:55.17 | leifmadsen | well StopMonitor() is required to stop recording, but it'll stop upon hangup too |
17:55.49 | leifmadsen | ok I'm off to start slow cooking a roast for dinner |
17:56.18 | paulc | hangup of the inbound channel though, right? so if I'm using multiple Dials to connect to different people, I should Monitor...StopMonitor around each call segment? |
17:56.26 | paulc | Mmm roast dinner.. sign me up! :) |
17:57.08 | leifmadsen | paulc: that's where it starts getting complicated (and don't even both thinking about transfers), so you'll have to experiment |
17:57.30 | paulc | LOL fair enough.. maybe the force be with me! :) |
17:57.30 | leifmadsen | remember that when you transfer the monitoring follows the channel where Monitor() (or MixMonitor()) were triggered from |
17:57.54 | leifmadsen | so when you transfer, it'll either be killed if the original channel hangs up, or it'll keep recording if you transfer the original channel |
17:58.07 | leifmadsen | A (Monitor) --> B --> transfer A --> C (call is recorded) |
17:58.25 | paulc | Transferring shouldn't be an issue here.. I think.. (interesting project: it's a phone sex type "which kind of kinky girl would you like to talk to" app.. bunch of CURL called PHP for back end stuff, + web interface, and "interesting" caller experience through the dialplan) |
17:58.26 | leifmadsen | A (Monitor) --> B --> transfer B --> C (A hangs up, call stops being recorded) |
17:59.02 | Katty | http://i.imgur.com/ZfGQX.jpg <- ^_________^ |
17:59.18 | paulc | Shouldn't be any transferring in this scenario I don't think... A (monitor) --> B.. either B hangs up or A presses *.. A (monitor) --> C.. rinse and repeat |
17:59.46 | paulc | Katty: I don't get it? (still feeling fuzzy - it's early!) |
17:59.51 | leifmadsen | paulc: that sounds easier :) |
18:00.04 | Katty | paulc: that is molecular structure of glucose |
18:00.05 | paulc | Katty: Ah.. chemical reference? |
18:00.05 | leifmadsen | ya I don't get it either |
18:00.10 | leifmadsen | oh! :) |
18:00.19 | anonymouz666 | leifmadsen: unless you detect the transfer and when you call the C you use M in Dial() to start MixMonitor again (in case that source hangup) |
18:00.20 | Katty | Sugar Cube |
18:00.22 | paulc | Talk about geeky number plates :) |
18:00.30 | Katty | i'd totally drive it |
18:00.38 | leifmadsen | anonymouz666: ya now you're getting into voodoo :) |
18:00.46 | leifmadsen | detecting transfers is hard |
18:01.07 | leifmadsen | who does the transfer and all that makes it so you don't always know you're being transferred (blind transfers for example) |
18:01.08 | *** join/#asterisk z4nD4R (~zandar@si-nat-61.ehs.sk) |
18:01.15 | z4nD4R | hi .. mery x-mass |
18:01.19 | leifmadsen | when you transfer, there is no dialplan being executed, just channels being bridged |
18:01.34 | leifmadsen | (at least not dialplan that knows, "hey you're going to transfer a call!") |
18:01.37 | z4nD4R | somebody to help with voicemail? |
18:01.47 | leifmadsen | ~ask |
18:01.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
18:01.53 | paulc | @leifmadsen: it's actually been a fun project so far.. especially when you use text to speech for dev prompts.. nothing like robotic Allison saying things like "If you're into Wild Women in Uniform, press 1..." (and that's one of the tamer options!) |
18:02.11 | paulc | Z4nD4R: What's up with voicemail? |
18:02.11 | leifmadsen | presses option 4 |
18:02.29 | paulc | Option 4 = Fetish Categories.. |
18:02.30 | leifmadsen | paulc: I'm building a psychic network right now... it's been a nightmare :) |
18:02.31 | paulc | you sure you want to go there |
18:02.39 | leifmadsen | presses * |
18:02.55 | paulc | To talk to our hottest fantasy women, press 1... |
18:03.05 | z4nD4R | paulc: i have set voicemail ... http://www.asteriskguru.com/tutorials/asterisk_voicemail.html ... but dont send me mail about voicemail |
18:03.09 | paulc | Psychic eh? Similar kind of stuff to this project then.. (maybe we should be collaborating!) |
18:03.15 | leifmadsen | I should have used a psychic to find out if this project would have had the call flow re-done 4 times |
18:03.18 | *** join/#asterisk AndyMLdroid (~andyml@32.138.125.4) |
18:03.39 | leifmadsen | ok, I gotta start this roast, so I'll be back in a bit :) |
18:04.03 | paulc | z4nD4R: You've got the email addresses in voicemail.conf? In the right place? (There's one for "pager" and one for "email"?). Other thing is, can you successfully send email from that box to yourself from the command line, outside of Asterisk? |
18:04.07 | *** join/#asterisk Russ (~russ@ip68-111-71-150.oc.oc.cox.net) |
18:04.18 | paulc | @leifmadsen: Cheers - see you in a bit :) |
18:05.29 | z4nD4R | paulc: dnot know how to send email from commandline.... but mail should by set right |
18:07.01 | z4nD4R | paulc: 71001 => passw ,name , mail@gmail.com ; my record in voicemail.conf |
18:09.41 | z4nD4R | paulc: you see som error? |
18:09.53 | paulc | z4nD4R: Yeah, something like: 71001 => 1234,John Smith,john@acme.com |
18:10.24 | paulc | z4nD4R: Your line looks ok to me. Have you restarted Asterisk since changing voicemail.conf? |
18:10.36 | z4nD4R | offcourse |
18:10.55 | paulc | z4ND4R: You really need to make sure mail works from the command line too. Something like "mail john@acme.com" then follow the prompts - ensure the mail reaches the destination. |
18:11.31 | z4nD4R | i started softphone.. and i becomme notification about voicemail |
18:12.02 | z4nD4R | paulc: how to make mail from command line? |
18:12.42 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
18:24.15 | Katty | http://i.imgur.com/rdifY.jpg <- so i was thinking dinner this weekened....andddd...i found this inspiring. perhaps it will inspire a few more ;) |
18:24.20 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
18:24.47 | Katty | but what should i put under the cheese? |
18:24.51 | Katty | thinking veggies. |
18:25.48 | Qwell | the answer is obvious |
18:25.50 | Qwell | more bacon! |
18:25.59 | *** join/#asterisk m_tadeu (~quassel@89-180-98-120.net.novis.pt) |
18:26.01 | Qwell | actually, ground beef probably would go well with that |
18:26.20 | thehar | BACON BACON BACON |
18:26.22 | wizard171 | Eggs, Potatoes ... |
18:26.41 | Qwell | wizard171: that would work too. breakfast...thing. |
18:26.50 | Qwell | bacorito |
18:27.00 | _Corey_ | looks healthy to me |
18:27.16 | Katty | dies |
18:27.19 | Katty | oh |
18:27.19 | Katty | egg |
18:27.21 | Katty | quiche, yes |
18:27.23 | Katty | i like this idea |
18:27.37 | Katty | upvotes Qwell, again |
18:28.28 | paulc | z4nD4R: delayed reply, sorry... type "mail john@acme.com" (using your email address) from the command line and see what happens |
18:29.04 | z4nD4R | paulc: command line form asterisk or from linux console? |
18:29.31 | paulc | z4nD4R: from the linux console - we're testing that your machine is physically capable of sending mail, before we figure out why Asterisk voicemail emails aren't being sent |
18:30.33 | z4nD4R | paulc: i dont have "mail" instaled... i can say it is pure install ... only asterisk is installed |
18:31.01 | leifmadsen | or install mutt |
18:31.04 | leifmadsen | which I find easier to use |
18:31.15 | Katty | also likes mutt |
18:31.31 | z4nD4R | leifmadsen: so i muss set in voicemail? |
18:31.39 | z4nD4R | voicemail.conf ? |
18:32.10 | leifmadsen | eh? |
18:32.18 | leifmadsen | you were talking about 'mail' command |
18:32.54 | z4nD4R | leifmadsen: yes.. but it is deeper :D why my asterisk dont send mail about voicemail |
18:33.25 | Katty | z4nD4R: have you confirmed it sends emails. period. |
18:33.47 | z4nD4R | Katty: ? |
18:34.54 | Katty | i'll take that as a no. |
18:35.13 | Katty | z4nD4R: you need to make sure your asterisk machine can properly send emails before working with the voicemail app. |
18:35.34 | Katty | z4nD4R: as the voicemail app is dependant on the server's ability to properly send mail. |
18:36.01 | z4nD4R | Katty: yes... but how??? which program? mail? mutt? which is use by asterisk? |
18:36.10 | leifmadsen | z4nD4R: do you have sendmail installed? does your provider block outbound relayed email? does the other end reject your email since you don't have an MX record setup? |
18:36.13 | leifmadsen | sendmail |
18:36.16 | leifmadsen | (or postfix) |
18:36.34 | leifmadsen | then you have to check /var/log/maillog to verify it is sending email |
18:37.09 | Katty | z4nD4R: i use mutt to do that. |
18:37.16 | Katty | z4nD4R: to test emails. |
18:37.22 | Katty | z4nD4R: mutt is simply an email client. |
18:37.24 | *** join/#asterisk bmg505 (~leon@196-209-7-245.dynamic.isadsl.co.za) |
18:37.41 | Katty | z4nD4R: you can send an email from command line using mutt, outside of asterisk. |
18:38.10 | z4nD4R | Katty: ok .. u have thi in voicemail? send mail with mutt? |
18:38.19 | Katty | z4nD4R: mutt is an email client. |
18:38.29 | Katty | z4nD4R: asterisk does not use mutt. |
18:38.44 | z4nD4R | Katty: what use asterisk to send emails? |
18:38.48 | Katty | z4nD4R: i believe asterisk uses sendmail. |
18:39.01 | Katty | z4nD4R: i use mutt to test from commandline |
18:39.02 | leifmadsen | <leifmadsen> z4nD4R: do you have sendmail installed? does your provider block outbound relayed email? does the other end reject your email since you don't have an MX record setup? |
18:39.29 | leifmadsen | question asked and answered |
18:39.46 | z4nD4R | ok w8 .. i go try it mutt and sendmail.. and report :) |
18:39.59 | Katty | mutts leifmadsen |
18:40.00 | leifmadsen | chooses not to wait, and goes to eat lunch |
18:40.08 | Katty | leifmadsen: enjoy. |
18:40.55 | paulc | I quite enjoy reading Kyle Rankin (or is it Shawn Powers?) go on about mutt in Linux Journal |
18:41.08 | paulc | I use it once in a while too.. |
18:41.15 | paulc | it's a fun day today :) |
18:42.02 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
18:42.02 | _Corey_ | forgot to renew his subscription to LJ... |
18:42.26 | _Corey_ | I got that thing for more than 10 years... good stuff |
18:44.00 | paulc | Yeah, it's a good read generally. That and Wired - my two favourites. |
18:44.32 | _Corey_ | The other one I used to stash was Dr Dobbs journal |
18:44.56 | _Corey_ | I don't write much code anymore but there are some good ideas in there |
18:47.52 | paulc | dear sox, telling me "pan is deprecated" is nice, but how about a hint of what to use instead? *grumbles* |
18:50.53 | z4nD4R | Katty: paulc 1. i can send email from linux console .. troght mutt |
18:51.42 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
18:52.11 | paulc | z4nD4R: using the same email address as in your /etc/asterisk/voicemail.conf, right? |
18:54.03 | z4nD4R | right |
18:55.33 | paulc | z4nD4R: Hmm.. how about voicemail contexts? you're leaving a message in the right mailbox/context that has the email address set up? |
18:55.54 | z4nD4R | but asterisk dont use mutt.... in voicemail.conf... isee mailcmd = sendmail -t ( bot comment thi line ) |
18:56.40 | Katty | feels like naptime |
18:57.53 | paulc | z4nD4R: so have you tested sending mail using sendmail? |
18:58.12 | z4nD4R | paulc: no...only using mutt |
18:58.27 | z4nD4R | if i understand.. i have 2 options |
18:58.39 | z4nD4R | set voicemail.conf to use mutt |
18:58.43 | z4nD4R | or install sendmail |
18:58.46 | z4nD4R | right? |
18:59.33 | paulc | z4nD4R: no.. because mutt is a user program for sending mail, whereas sendmail is a mail transport.. MUA versus MTA or something |
18:59.37 | paulc | mutt was just to test if mail went through |
18:59.44 | paulc | it did, but we don't know if it used sendmail or went direct etc |
19:00.01 | paulc | from the commandline, can you do "mail john@acme.com" or "sendmail -t john@acme.com" |
19:00.03 | z4nD4R | so i muss install sendmail.. |
19:00.18 | paulc | if you don't have sendmail installed, then yes. |
19:00.36 | z4nD4R | ok |
19:00.37 | Qwell | I would be shocked if you didn't have sendmail. |
19:01.23 | makafre | To what does the Resource field correspond when we do jabber show buddies ? Its right under the Buddy name... |
19:01.38 | z4nD4R | paulc: ok installed |
19:01.51 | Qwell | makafre: it means nothing. it's just a (user-configurable) identifier |
19:02.05 | Qwell | people use Home, Work, Tacos, etc |
19:02.19 | z4nD4R | paulc: i need some setting to sendmail? |
19:02.27 | makafre | Qwell, ok, I am asking because when I try to get JabberStatus it complains about the resource being null |
19:02.39 | Qwell | null is probably not valid |
19:02.58 | Qwell | I have no idea how to set/use it in Asterisk. |
19:03.02 | paulc | z4nD4R: I don't understand the question |
19:03.10 | makafre | ok |
19:05.23 | makafre | about the sendmail thing, I always use sendEmail, it s a script and its easy to use, I prefer it over sendmail |
19:06.00 | makafre | here is link to get it: http://caspian.dotconf.net/menu/Software/SendEmail/sendEmail-v1.56.tar.gz |
19:06.23 | z4nD4R | paulc: i have installed sendmail.... some additional options??in voicemail.conf.. or? |
19:06.24 | Qwell | Asterisk doesn't need easy |
19:06.33 | Qwell | z4nD4R: No, sendmail requires no configuration. |
19:06.58 | z4nD4R | Qwell: ok i try it sendmail -t my@email.com |
19:07.06 | z4nD4R | but nothing to do |
19:09.07 | z4nD4R | Qwell: but in voicemail.conf i have commented line ;mailcmd = /usr/bin/sendmail -t .. uncommented this? |
19:09.21 | Qwell | No. Do nothing. Leave a voicemail, and it will just work. |
19:09.40 | citywok | s/will/should/ |
19:09.59 | Qwell | Do not question The Qwell! |
19:10.11 | citywok | I didn't question the qwell. I just corrected him :P |
19:10.17 | Qwell | oh snap |
19:10.35 | citywok | lmao |
19:10.50 | citywok | good thing russellb isn't here or i might have gotten KB'd for that |
19:10.55 | z4nD4R | Qwell: but not work :D |
19:11.19 | z4nD4R | Qwell: sry... it works :D |
19:11.33 | Qwell | citywok: see? :P |
19:11.44 | citywok | hahaha, no i didnt :P |
19:11.51 | citywok | nothing to see here, please move along |
19:12.09 | Qwell | z4nD4R: so, fixed? |
19:13.05 | z4nD4R | Qwell: yes.. i becomme mail with text and the voice messenge.. :) thx.. i muss only edit the text.. and it is ok.. |
19:16.55 | Qwell | I'm not sure what that means, but alright. |
19:17.22 | z4nD4R | Qwell: text which is send in mail |
19:17.39 | Qwell | oh, okay. thanks for clearing that up. |
19:19.19 | *** join/#asterisk jetlag (jetlag@pool-173-61-245-217.cmdnnj.east.verizon.net) |
19:27.28 | makafre | how to call a dialplan function from a perl AGI? |
19:27.48 | Qwell | same way you would use a variable |
19:27.56 | makafre | ok, i see, thanks |
19:28.12 | *** join/#asterisk Corydon76-home (eleven@c-69-137-80-31.hsd1.tn.comcast.net) |
19:28.12 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
19:28.30 | *** join/#asterisk Corydon76-home (six@c-69-137-80-31.hsd1.tn.comcast.net) |
19:28.30 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
19:37.06 | makafre | anyone knows who is the sender parameter for the JABBER_STATUS function? (ref: http://www.asterisk.org/docs/asterisk/trunk/functions/jabber_status) |
19:37.49 | makafre | i know the resource and buddy |
19:38.16 | makafre | but the sender? |
19:38.33 | Qwell | that would be you |
19:38.44 | makafre | me? |
19:38.56 | Qwell | your account |
19:39.18 | Qwell | You can have multiple accounts on a box, so you need to specify who you want to send from. |
19:39.18 | makafre | ok, isn't it the same as the resource? I mixed up |
19:40.19 | makafre | let's say I have in jabber.conf [tacos] connecting as username=admin, which one is the resource and which one is the sender? |
19:44.54 | makafre | ok, I guess you mean they can be the same if there is only one user defined |
19:46.43 | Qwell | I have no idea what resource is for, in that context. |
19:47.45 | makafre | ok, sure |
19:48.33 | *** join/#asterisk oDesk (~f@188.53.70.113) |
19:49.01 | makafre | well, it doesn't work with tacos, <buddy> |
19:49.04 | makafre | lol |
19:49.38 | oDesk | i want to Set(CALLERID(name)=D63xxxxxx) cut the D letter into this line where it show sets the caller ID |
19:50.33 | p3nguin | resource: |
19:50.34 | p3nguin | Client or transport Asterisk users to connect to Jabber. |
19:51.11 | *** join/#asterisk luckman212 (luckman212@pool-96-246-172-198.nwrknj.fios.verizon.net) |
19:51.59 | p3nguin | It makes sense to me that the resource is NOT the username. |
19:52.04 | makafre | ok, so If I have [tacos] username=admin@xxxxxxx.com "tacos" would be the resource |
19:52.37 | p3nguin | Since resource isn't the username, there aren't too many other choices. |
19:52.49 | Katty | hi |
19:53.00 | p3nguin | hi2u |
19:53.01 | Qwell | Katty: ohai |
19:53.10 | Katty | Qwell: your turn for napping |
19:53.29 | makafre | I tried with admin@xxx.com, buddy@xxx.com, tacos but it doesnt work, unless I have a bug in my AGI |
19:54.24 | wizard171 | No, "username=admin@host/resource" where "resource" = something like "home" vs "work" etc. |
19:54.59 | makafre | ah |
19:55.10 | p3nguin | I often use my host's name as the resource on my client. |
19:55.31 | p3nguin | I think Psi does that by default. |
19:56.12 | wizard171 | ah, but do your ... resources ... need hosting? (that is the question?) eh? |
19:56.34 | p3nguin | I have no idea what you're even trying to say. |
19:56.41 | wizard171 | ;) |
19:56.58 | makafre | let me try that |
19:57.56 | *** join/#asterisk ickmund (~ickmund@c-9149e755.015-144-70697410.cust.bredbandsbolaget.se) |
19:58.01 | makafre | question, are we talking about the username's resource or the buddy's resource? |
19:58.27 | p3nguin | A very good question that is. |
19:58.34 | makafre | lol |
19:59.02 | p3nguin | I'm not even sure what you're trying to do. |
19:59.51 | makafre | I am trying to use JABBER_STATUS in order to get the status and make decision base on it. I only have one asterisk and one jabber server in my environment |
20:00.33 | makafre | I was able to do it using JabberStatus but since it's deprecated I wanted to go with the function instead |
20:02.10 | Katty | someone lend me 75 cents |
20:02.13 | Katty | i need caffeine |
20:02.21 | leifmadsen | tosses Katty a quarter |
20:02.49 | Katty | leifmadsen: i'm about to run to the store and buy a case of mt dew |
20:02.56 | leifmadsen | well done |
20:03.00 | thehar | excellent |
20:03.08 | Katty | hide them in the bottom drawer of the fridge |
20:03.17 | Katty | no one else looks down there in the veggie drawer but me ^___________________________^ |
20:03.17 | thehar | nice |
20:03.31 | thehar | nom |
20:05.02 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
20:05.10 | leifmadsen | makafre: show us how you're using JABBER_STATUS() |
20:05.24 | makafre | ok, give me a minute |
20:05.43 | Katty | wonders |
20:05.49 | thehar | if |
20:05.52 | Katty | if i can nap on the clock, i should be able to support local economy on the clock |
20:05.55 | Katty | right?? |
20:06.00 | leifmadsen | Set(JabberResult=${JABBER_STATUS(lmentinc,pbx@leifmadsen.com)}) |
20:06.06 | p3nguin | We has no one configured online payments for soda machines yet? I could paypal your vending machine and you could go pick up your drink. |
20:06.24 | Katty | p3nguin: that, sir, is amazing idea. |
20:06.27 | leifmadsen | makafre: there is an example |
20:06.37 | p3nguin | s/We/Why/ |
20:07.09 | citywok | hmm. is chan_gtalk still allowing calls to PSTN? |
20:07.18 | Katty | i bet i could paypal my company |
20:07.23 | Katty | and they'd get me a soda ^_^ |
20:08.40 | Katty | tada! |
20:08.42 | Katty | afks |
20:08.49 | leifmadsen | citywok: ya as long as you update to 1.8.1.1 or later |
20:09.01 | citywok | hmm. okay i've got 1.8.1.1, then that didnt like me so i went to trunk. |
20:09.07 | leifmadsen | odd |
20:09.13 | leifmadsen | following the docs? |
20:09.44 | citywok | some random tutorial i google foundified |
20:09.59 | makafre | here it is: http://pastebin.ca/2027523 |
20:11.16 | makafre | basically I call extension 10 which sends the "Incoming call" text to Spark, and then I am trying to send the status the same way |
20:11.19 | citywok | attempting the wiki.aster now, i was 95% of that |
20:13.34 | leifmadsen | citywok: make sure you use the one on the asterisk wiki |
20:13.39 | leifmadsen | ah, you said that |
20:13.57 | leifmadsen | ya the asterisk wiki is going to be the most up to date (Malcolm keeps it pretty up to date) |
20:14.06 | Katty | mmmm, caffeineee |
20:16.27 | Katty | has anyone planned christmas dinner yet? |
20:16.39 | leifmadsen | I plan to eat whatever my mom makes :) |
20:16.54 | Qwell | Katty: whatever you make and then bring me :p |
20:16.55 | WIMPy | Excellent choice :-) |
20:18.01 | Katty | Qwell: candied ham, scalloped potatos, broccoli |
20:19.23 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
20:21.01 | makafre | leifmadsen: so you have [lmentinc] username=pbx@leifmadsen.com in your jabber.conf |
20:25.48 | leifmadsen | makafre: no, I have [lmentinc] defined and pbx@leifmadsen.com is the jabber user I'm checking the status of |
20:26.22 | makafre | ok |
20:27.48 | *** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au) |
20:35.54 | makafre | grrrr, problem solved, code 13 |
20:35.58 | makafre | lol |
20:36.15 | Katty | i'll code your 13 in a minute. |
20:36.25 | makafre | haha |
20:37.11 | WIMPy | Do we hace to do metric conversions on error codes now? |
20:38.00 | makafre | my problem was in the way I was calling get_variable |
20:38.32 | makafre | now is that moutain dew still available ;-) |
20:38.41 | Katty | it's in the veggie drawer. |
20:41.15 | makafre | get_moutaindew("veggie drawer"); |
20:43.31 | *** join/#asterisk riscphree (~riscphree@h235.10.91.75.dynamic.ip.windstream.net) |
20:43.33 | Katty | carrar: ooooooooh |
20:43.34 | Katty | carrar: btw |
20:43.39 | Katty | carrar: got your card ^_^ |
20:44.02 | Katty | carrar: it's on my fridge. |
20:45.24 | p3nguin | is suddenly reminded to check the mail |
20:47.54 | *** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au) |
20:51.52 | CaneToad | I am registering asterisk with an external provider and am using /[extension] as part of the registration so that incoming calls are forwarded to that extension...and incoming calls work, but there's something different from when registering the ATA directly with the provider...the phone displays caller id but alternates between the extension number and the caller id, and the phone's distinctive ring for certain numbers does |
20:52.41 | CaneToad | the phone is displaying the correct caller id, but unlike when registering with the provider directly, somehow the extension number is displayed too |
20:55.28 | WIMPy | Has anyone here ever had any success with DAHDIs inbanddisconnect=yes? I only get random audio. |
20:57.26 | WIMPy | With random=either of busy, silence or a dialtone. |
20:57.42 | ChannelZ | Darn. I was hoping you meant showtunes and rap |
20:59.13 | Katty | hi ChannelZ |
20:59.16 | Katty | hugs ChannelZ |
20:59.25 | ChannelZ | Hey Katty! LTNS |
20:59.28 | ChannelZ | Merry Christmas |
20:59.33 | Katty | ty (= |
20:59.38 | Katty | what you been up to |
20:59.57 | ChannelZ | Nothing too exciting, just keeping busy with work |
21:00.00 | ChannelZ | You? |
21:00.16 | Katty | going through tons of wrapping paper |
21:00.21 | Katty | but it's going to be so incredibly worth it |
21:00.40 | ChannelZ | heh yeah that's one of my many jobs later tonight |
21:00.46 | Katty | close friend of mine has been going through a very difficult time lately. single mother, living at home with her father. |
21:00.54 | Katty | working to put herself through college |
21:01.00 | ChannelZ | Gotta finish cleaning the house, decorating, have a cheesecake to make... wrap... |
21:01.12 | Qwell | I want a cheesecake. |
21:01.17 | Katty | they don't have a lot of money, so i have decided to spoil the be-jeesus out of her son |
21:01.25 | ChannelZ | :) that's cool |
21:01.43 | Katty | i think i'll take my camera over and make a video of it |
21:01.53 | ChannelZ | Qwell: unless you live in Colorado, you're out of luck :) |
21:02.03 | Katty | mm cheesecake. |
21:02.05 | ChannelZ | Well.. not completely, but at least of mine. |
21:02.13 | Katty | my cheesecake recipe is so easy... |
21:02.16 | Katty | no baking required |
21:02.24 | Katty | it's on the back of the gelatin box |
21:02.33 | Katty | don't tell tho, everyone thinks its amazing ;) |
21:02.39 | Qwell | jello cheesecake? |
21:02.45 | Katty | no, gelatin packets. |
21:02.45 | ChannelZ | I've had one of those a long time ago I think |
21:02.57 | Katty | Qwell: it's in the baking aisle. |
21:03.14 | Qwell | does cheesecake usually use gelatin? |
21:03.23 | ChannelZ | No |
21:03.51 | Katty | usually you bake it in a spring form pan |
21:03.52 | ChannelZ | Speaking of food I need to go get lunch. BBL |
21:04.08 | Katty | but the knox gelatin has a recipe on the back where you through 3 or 4 things in the blender with cream cheese and gelatin |
21:04.12 | Katty | then chill it. |
21:04.15 | Katty | POOF insta-cheesecake |
21:04.26 | Katty | s/through/throw/ |
21:10.16 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
21:11.24 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.182.59) |
21:13.20 | GhOnDiE | hi people i have my asterisk setup on an internal network and would like to have some remote extensions setup, the config is done and all ports have been forwarded direct to the asterisk box, however the remote clients refuse to register, and when they have done i can only make calls direct to other sip clients and not dial feature codes |
21:13.29 | Katty | aww. |
21:13.32 | Katty | my boss told me to go home |
21:13.38 | GhOnDiE | when looking at the sip debug it comes up with a 401 error |
21:13.56 | GhOnDiE | im guessing its a NAT problem but im not sure how to solve it |
21:13.58 | z4nD4R | Katty: happy man :D |
21:14.42 | Katty | z4nD4R: he can be |
21:14.50 | z4nD4R | Katty: :D |
21:14.56 | Qwell | Katty: Go home! |
21:15.23 | Katty | i'm almost done |
21:16.02 | Katty | what is the asterisk abbreviation for thursday |
21:16.40 | Katty | thu? |
21:16.57 | Qwell | probably |
21:17.23 | Katty | let's hope so! |
21:18.07 | Katty | well it didn't squeal at me. |
21:18.27 | Katty | hope everyone has an awesome weekend if i don't see you before monday (= |
21:18.31 | Katty | Qwell: especially you |
21:18.41 | Qwell | <3 |
21:18.50 | wizard171 | Katty: Merry Christmas ... ! |
21:35.49 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
21:37.48 | tyman | Anyone here figured a way to use CMU Sphinx (speech-to-text) with asterisk? |
21:38.37 | _Corey_ | tyman: There are some howtos out ther |
21:38.46 | _Corey_ | s/ther/there/ |
21:39.00 | _Corey_ | It's a little rough though compared to Lumenvox |
21:39.15 | _Corey_ | unless commercially prohibitive, I'd recommend Lumenvox |
21:40.00 | tyman | _Corey: I wasn't aware that Lumenvox did voice-to-text transcription? Thought only recognition. |
21:40.18 | _Corey_ | oh, sorry, I thought you were talking about speech rec |
21:40.31 | paulc | I've played with Lumenvox and really liked it.. speech rec auto attendant type deal.. the fun bit was the thresholds.. "I think you said New York.. am I right?" |
21:40.41 | _Corey_ | lol, yeah |
21:40.45 | _Corey_ | some words are dead-on |
21:40.49 | _Corey_ | others, not os |
21:40.51 | _Corey_ | er so |
21:41.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
21:41.42 | tyman | _Corey: Yeah, lumenvox looks cool. Heard good things about it. Just looking to do the gvoice type email/jabber of vmail |
21:42.21 | paulc | speaking of jabber.. I've been playing with that for my current project.. and Asterisk seems to get a bit bent out of shape if it can't find people on it's roster |
21:42.49 | tyman | easy enough sounding to do...just wondering people's experiences with CMU Sphinx or if there are better options |
21:42.52 | _Corey_ | If sphinx will do it with a WAV file or something, then it's just a dial plan or shell hack to do the job |
21:43.00 | paulc | like right now the "asterisk" jabber user has A, B and C on it's roster... I add D to the jabber server, the groups mean everyone sees everyone else, but Asterisk doesn't know about D until a reload |
21:43.08 | _Corey_ | i wasn't aware sphinx could do speech to txt |
21:43.38 | _Corey_ | I know of several commercial offers that are SaaS |
21:44.00 | tyman | http://nsh.nexiwave.com/2010/09/voicemail-transcription-with.html |
21:45.12 | _Corey_ | tyman: looks like you answered your own question... :) |
21:45.31 | *** part/#asterisk z4nD4R (~zandar@si-nat-61.ehs.sk) |
21:45.52 | _Corey_ | Looks cool |
21:46.03 | tyman | couldn't be much worse than gvoice's transcription accuracy! :-) I have some absolutely horrible, the point of laughable, transcriptions from gvoice. |
21:46.32 | _Corey_ | seriously, I use several as examples of how impossible accurate transcription is with clients |
21:46.46 | tyman | _Corey: looking to see if there are better options before I invest time in it |
21:47.12 | _Corey_ | google has the benefit of millions of goog-411 calls and hundreds of thousands of hours of training and it's still s*** |
21:47.12 | tyman | yeah, more of a novelty in most cases... |
21:47.46 | _Corey_ | but JUST accurate enough to be of utility |
21:48.09 | tyman | goog-411 is gone now...that didn't work to well for me for some reason... |
21:48.15 | tyman | s/to/too/ |
21:48.58 | p3nguin | I have more trouble with Bing 411 than I ever did with Goog-411. |
21:49.05 | _Corey_ | they must have decided they got all the training they needed |
21:49.32 | _Corey_ | I'm definitely going to poke around with PocketSphinx though, even if it's not as good as google, that's good enough for many customers |
21:49.45 | ChannelZ | WTF? You can only buy $50 max gift codes from iTunes? |
21:50.35 | _Corey_ | ChannelZ: Can't beat Amazon mp3 store |
21:51.07 | tyman | _Corey: have had pretty pathetic situations where i was playing phone tag with someone who had my gvoice number and kept leaving me vmail which was being text'ed to my iphone. I couldn't play them back so i'm desperately trying to decipher the transcriptions. No callback number... complete suckfest. :-) |
21:51.31 | thehar | perks up |
21:51.32 | thehar | what? |
21:52.04 | tyman | ChannelZ: Amazon mp3...only way to go. 256kbps vbr, with no horrible DRM |
21:52.07 | ChannelZ | _Corey_: yeah my personal choice but I was getting a couple gifts for clients who are in more or less apple fanboys |
21:52.14 | thehar | Katty: so raise your glass if you are wrong! in all the right ways! |
21:52.37 | _Corey_ | tyman: yeah, on their own transcriptions are totally unreliable |
21:52.46 | ChannelZ | tyman: thankfully iTunes has mostly given up on DRM but regardless I try to not personally buy anything from them |
21:53.36 | tyman | ChannelZ: shows you how long it's been since I bought songs from itunes (re: drm). :-) |
21:53.38 | _Corey_ | Prices are much better on Amazon too. |
22:05.25 | *** join/#asterisk sthon (~sean@fbx.caras.modwest.com) |
22:05.32 | *** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net) |
22:05.55 | sthon | is there any benefit to having hardware echo canceller? |
22:06.28 | *** join/#asterisk infratel (infratel@154.5.144.132) |
22:07.28 | ChannelZ | if you have lots of channels needing it at once, sure |
22:09.03 | jShaf | installed asterisk (for the first time) on mac (via homebrew) |
22:09.05 | jShaf | ./configure --prefix=/usr/local/Cellar/asterisk/1.6.1.6 --localstatedir=/usr/local/var --sysconfdir=/usr/local/etc --host=x86_64-darwin |
22:09.11 | WIMPy | They might have a better algorithm and they save CPU time. |
22:09.15 | jShaf | and yet i dont see any file in /usr/local/etc/asterisk/ |
22:09.15 | sthon | ChannelZ: I get this in dmesg a few times a week zaptel Disabled echo canceller because of tone (rx) on channel 25. Just wondering if it's worth getting a hardware echo canceller card. |
22:09.18 | jShaf | should there be one? |
22:09.20 | infratel | quick question about interface out avaibility, If I dial out a matching context and that interface, does this two line dialplan drop the out call to the second line/group? exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
22:09.20 | infratel | exten => _91NXXNXXXXXX,2,Dial(Zap/g2/${EXTEN:1}) |
22:09.20 | infratel | exten => _91NXXNXXXXXX,3,Congestion |
22:09.34 | infratel | ? |
22:09.52 | tyman | jShaf: did you make, make install? |
22:09.59 | _Corey_ | sthon: I've found the quality of the HW echo cans better than software alone |
22:10.03 | infratel | g1 drops the out call to g2 out? |
22:10.24 | sthon | _Corey_: Thanks. |
22:10.31 | _Corey_ | sthon: the tone thing is probably a fax |
22:10.33 | jShaf | tyman: I see that the homebrew script does make && make install |
22:10.38 | WIMPy | infratel: You should check DIALSTATUS and/or HANGUPCAUSE. |
22:10.47 | _Corey_ | sthon: when a fax tone is detected, the echo can automatically should disable |
22:11.15 | sthon | _Corey_: that's probably the case, it doesn't happen that frequently. |
22:11.18 | infratel | check dialstatus? a ordinary user/client would not be required to do that |
22:11.36 | tyman | jShaf: if there were errors with make then make install may not run...Did you have build errors? |
22:12.09 | WIMPy | infratel: ... in your dialplan. Befor trying again on the next interface. |
22:13.13 | infratel | Wimpy, its actually not my dialplan. Just a web site does not explain the grouping of two itentical dial plan lines with seperate group numbers. |
22:13.47 | infratel | It does not explain it so, my guess if zap out g1 is busy, then drop to the next line out zap2 |
22:17.08 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
22:18.11 | jShaf | tyman: I just now re-run the script, this time with verbose switch |
22:18.16 | jShaf | I dont see any error |
22:18.33 | jShaf | but now I wonder if it is because I did not run "make sample" ? |
22:18.38 | jShaf | thus no config files? |
22:23.07 | WIMPy | infratel: It will always try to dial out on group 2 if it faild on group 1, no matter what reason. |
22:27.11 | *** join/#asterisk Tili (~Tili@cm161.eta193.maxonline.com.sg) |
22:30.42 | infratel | okay that was the answer I was looking for. |
22:31.16 | infratel | Call it cascade channel aviability |
22:31.25 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
22:32.14 | WIMPy | That's why I suggested chacking DIALSTATUS and/or CAUSECOSE. |
22:32.55 | WIMPy | caause=34 would seem obvious. Not sure it there are others. |
22:33.24 | WIMPy | Maybe that would be a nice thing for Asterisk 1.10: CAUSELOCATION |
22:34.29 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
22:41.35 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
22:41.59 | shmaltz | why am I getting: load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc |
22:44.07 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
22:44.14 | ChannelZ | Did you build * yourself or is it from a package? |
22:44.34 | shmaltz | ChannelZ, myself |
22:45.16 | ChannelZ | hmnm. All of it at the same time? |
22:45.28 | shmaltz | ChannelZ, all of what? |
22:46.01 | ChannelZ | Asterisk. Like you didn't compile it and then a month later compile a part of it for a different module or something |
22:46.52 | shmaltz | nope, all at the same time |
22:47.01 | shmaltz | make, make install, make configs |
22:47.19 | ChannelZ | hmm |
22:48.40 | shmaltz | should I recomplie just that folder? |
22:49.41 | wizard171 | shmaltz: sounds like you are missing "autoconf" as a package during build, or you do not have "libcap.so"? (which, would be very strange) |
22:50.10 | ChannelZ | yeah or has something changed? (did it ever work?) |
22:50.21 | shmaltz | ChannelZ, I doubt it ever worked on this machine |
22:50.40 | shmaltz | wizard171, is it possible to compile anything without autoconf? |
22:50.46 | shmaltz | checking now for libcap.so |
22:51.02 | shmaltz | is libcap.so part of linux or asterisk? |
22:51.23 | ChannelZ | Yeah I'd figure if you were missing something it'd have just failed. Maybe your libcap is a crazy old version or... I dunno |
22:51.31 | shmaltz | root@pbx:/# ls -R | grep libcap.so returns some results |
22:51.34 | ChannelZ | it's separate |
22:51.59 | shmaltz | libcap.so verison is 2.19 |
22:53.12 | ChannelZ | hmmm |
22:54.26 | wizard171 | shmaltz: yes, you can compile lots without "autoconf" but, I have never tried asterisk without it ... and it sounds like that is what you are missing ... you may be missing other requirements as well ... bison/flex ... come to mind ... |
22:54.29 | shmaltz | it appears taht I'm getting on a few modlues |
22:54.44 | shmaltz | I'm going to reinstall libcap.so |
22:54.51 | ChannelZ | do you have libcap-dev or whatever your distro might call dev packages |
22:56.00 | wizard171 | ah, "configure" needs "autoconf" to generate "makefiles" ... so things link correctly ... if libcap.so is there, the issue is probably that its not being "linked" to ... |
22:56.18 | shmaltz | I'm running 1.4 |
22:56.24 | wizard171 | s/makefiles/otherstuff/ |
22:57.05 | shmaltz | wizard171, how do I "link" to it? |
22:57.50 | wizard171 | ordinarily, you don't ... the configure/make/make install will do it for you ... if all the right tools are installed ... autoconf,make,libtool,flex,bison, etc. ... |
22:58.00 | ChannelZ | heh interesting, looking through one of my config.log from a previous build, the lcap test failed |
22:59.28 | shmaltz | how do I know if I have bison/flex? |
22:59.55 | shmaltz | config.log reports both bison and flex |
23:00.43 | NightMonkey | Gosh I heart asterisk. |
23:01.07 | NightMonkey | I just moved my home phone system off a troublesome power-hungry server to a Sheevaplug. :) |
23:02.11 | shmaltz | an identical config (other than kernel) is running music on hold fine |
23:03.25 | wizard171 | libcap is part of "posix" ... and res_musiconhold ... forks ... perhaps your "posix" is one of the ... other implementations? |
23:03.29 | shmaltz | what is the name of the tool that relinks lib? |
23:04.02 | wizard171 | "ld" |
23:05.04 | shmaltz | didnt help :'( |
23:05.11 | shmaltz | :o |
23:05.41 | wizard171 | shmaltz: the tool you are looking for is "autoconf" in your "package" selections ... I can't seem to find a reference to what you are building on ? |
23:05.55 | shmaltz | slackware 13.1 |
23:06.15 | wizard171 | and you are using "yum" or ??? |
23:06.27 | shmaltz | pkgtool |
23:07.50 | shmaltz | is recompiling from scratch |
23:10.01 | shmaltz | recompiling didnt help :'( |
23:11.40 | wizard171 | grab the latest "autoconf" from "ftp.gnu.org" and install it, then try it again ... |
23:11.43 | shmaltz | could it be a kernel options that I'm missing? |
23:12.32 | *** join/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp) |
23:12.49 | dongs | hi, im using asterisknow i need to add a new IP to the machine wiht it, how do i do that. |
23:13.27 | p3nguin | This is the Asterisk channel, not the AsteriskNOW channel. |
23:14.15 | p3nguin | In here, we add IP addresses to interfaces with 'ip' or 'ifconfig'. |
23:14.21 | dongs | yes, it also has 195 people (not counting you as a person) and the other one has 11 bots. |
23:14.35 | dongs | so asking here is much more likely to get a response |
23:15.01 | dongs | p3nguin: thats great, if I knew the login/pass to whatever OS is behind asterisknow i'd do it there too. |
23:15.09 | *** join/#asterisk mpe (~mpe@0x5581329f.terminal.tdcmobil.dk) |
23:15.34 | p3nguin | It's CentOS, and all the pertinent information is found in the AsteriskNOW quickstart guide. |
23:16.51 | p3nguin | I'd be somewhat surprised if you didn't have to set the root password during setup. |
23:17.17 | shmaltz | wizard171, I did that but autoconf -V still returns 2.65 when I just did make install on 2.68 |
23:19.01 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
23:19.44 | shmaltz | dongs, just boot to userlevel 1 and change it |
23:19.55 | shmaltz | or dual boot and chroot and run passwd |
23:20.26 | dongs | yeah, i did remember setting password and my default one worked. |
23:20.33 | dongs | interface added, problem solved |
23:20.39 | dongs | now i can go ask freepbx guys all stupid questions. |
23:20.50 | shmaltz | dongs, :-h |
23:21.34 | dongs | thanks |
23:21.35 | *** part/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp) |
23:28.04 | wizard171 | shmaltz: strange ... in the source directory, for asterisk, you may need to "make clean" ... then "configure/make" again ... |
23:28.39 | shmaltz | wizard171, I deleted the whole directory before recomplining |
23:28.43 | shmaltz | does that count? |
23:29.13 | wizard171 | :) ... yep! (but, my brain is working on the possibles??) |
23:29.30 | shmaltz | ok, so I should still do make clean? |
23:30.11 | wizard171 | Not if "autoconf" was there ... before ... "configure/make" ... |
23:30.27 | wizard171 | on a fresh directory ... |
23:31.03 | shmaltz | wizard171, restarting the machine, lets see if that helps (i doubt it, but trying) |
23:31.30 | shmaltz | is rooting a nookcolor at the same time |
23:31.59 | wizard171 | okie, dokie, then ... sometimes I hear that works on ... windows ... as well ... ;) |
23:32.33 | shmaltz | ok restart gave me autconf 2.68 |
23:32.40 | shmaltz | lol |
23:33.33 | shmaltz | now recomplining |
23:33.55 | wizard171 | *that* should be ... interesting ... |
23:34.30 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
23:36.20 | shmaltz | still no good :'( |
23:39.03 | shmaltz | wizard you think asterisk 1.4.38 will be better than 1.4.37? |
23:39.41 | wizard171 | ah, some reason you wanted 1.4 ... vs any other ? |
23:40.13 | shmaltz | wizard171, like 1.2? |
23:40.35 | wizard171 | heaven forbid! ... how about 1.6.2.13? |
23:41.04 | shmaltz | wizard171, dont push it, I have been using 1.2 till like 3-4 weeks ago, why jump so high? |
23:41.37 | shmaltz | most of my installs are STILL running 1.2 |
23:41.46 | shmaltz | i might still have one 1.0x somewhere |
23:42.16 | wizard171 | you just *have* to have goals ... :) ... however, 1.4 is what you want, okay then, no offense meant ... :) |
23:42.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
23:42.44 | shmaltz | :) |
23:46.04 | shmaltz | of course 1.4.38 didn't help |
23:46.17 | shmaltz | ok, posting to lists |
23:46.45 | wizard171 | ah, how bout a "pastebin" of the "logs" from "configure/make" ...? |
23:47.00 | shmaltz | ok, my sis in-law just emailed me this: |
23:47.02 | shmaltz | Dear Santa..,, My wish for this year is a big fat bank account & a thin body. Please don't mix these 2 up like you did last year...<:p |
23:47.14 | shmaltz | wizard171, is that config.log? |
23:47.36 | wizard171 | yes ... |
23:49.14 | paulc | shmaltz: your sister makes me laugh - I can relate :) |
23:49.47 | WIMPy | Sister in alaw or sister in ulaw? |
23:49.53 | shmaltz | :D |
23:49.56 | paulc | ChanSpy - you're spying and the call ends. ChanSpy just sits there doing nothing. This is right, right? Because it's waiting for you to press * to skip to the next or whatever? |
23:50.04 | paulc | WIMPy: *groan* that's awful! |
23:51.41 | wizard171 | paulc: yes, unfortunately, until 1.8 ... then you can give arg to have it do just one and exit ... |
23:56.30 | wizard171 | shmaltz: I am not abandoning you ... I just need to go AFK for a few ... |
23:56.44 | shmaltz | thanks |
23:56.49 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
23:57.05 | shmaltz | wizard171, here is the pb: http://pastebin.ca/2027732 |