IRC log for #asterisk on 20101223

00:05.48bmoraca_workwhy do telephone companies suck so much to deal with?  uhg
00:09.49*** join/#asterisk erinspice (~erin@207.98.195.107)
00:11.16WIMPyI'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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00:17.06radenbmoraca_work, u tell me
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00:29.20*** join/#asterisk ldiamond (46331fee@gateway/web/freenode/ip.70.51.31.238)
00:29.56ldiamondOne 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)
00:30.07ldiamondAnyone have an idea how to solve such an issue?
00:30.52WIMPyGet a better connection (or allow a bigger lag).
00:31.08ldiamondWhere can we allow bigger lag?
00:31.20ldiamondas for the connection... well it's 3G... what can I do?
00:35.59WIMPyqualify=<ms>
00:36.03*** part/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com)
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00:38.05devmodUhm asterisk is running but it suddenly stopped listening on port 5060. Any idea hwo to debug that issue?
00:38.45WIMPydevmod: Read that before, but it hasn't happened to me.
00:39.08devmodIt's really weird. hmm
00:39.45WIMPyUgh. Dahdi sending shit again :-(
00:41.12wizard171WIMPy: 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?
00:42.04WIMPywizard171: No, it only dies so when running dahdi_cfg. Thereafter everything is fine.
00:42.11WIMPys/dies/does/
00:42.50WIMPyVoice quality is flawless.
00:43.36wizard171no ticking, popping, or sharp snapping noises of any kind?
00:44.47WIMPynope
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00:45.27wizard171devmod: ah, does "module show like sip" still show ... something ?
00:49.39devmodwizard171, I killed it :/ I had forgotten to enable any kind of logging
00:53.51wizard171devmod: that ... would make debugging ... hard ... ;)
00:54.49wizard171WIMPy: I am wading thru "userspace" ... if I go to sleep, poke me ... :)
00:54.54devmodwizard171, right, I wasnt expecting sip to just die :P
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00:55.27wizard171devmod: :) ... no one ever does ... until it does ... eh?
00:55.33WIMPywizard171: I wouldn't. Sleep ist important!
00:55.42devmodhaha it just crashed again
00:55.51devmodi mean , sip crashed
00:56.27devmodchan_sip.so                    Session Initiation Protocol (SIP)        2
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01:01.24wizard171WIMPy: did you build the kernel yourself? (ie: did you "enable userspace")
01:02.12WIMPywizard171: I did. What option/feature is that?
01:02.48WIMPyIguessed thare's something missing, I just couldn't find out, what.
01:05.07wizard171WIMPy: in udev or hotplug ... I am looking for exactly where ... which kernel are you using ?
01:05.26WIMPyI don't have udev running.
01:05.35WIMPy2.6.36.2
01:06.27wizard171WIMPy: ah, unless I am mistaken, udev is required for hotplug, and loading of firmware from "userspace" ... can you start it ?
01:06.41WIMPyOH.
01:09.20WIMPywct4xxp 0000:03:08.0: VPM450: Present and operational servicing 2 span(s)
01:09.59wizard171Yeah!  Congrats!
01:10.07WIMPyI thought that were two differen things, as no hardware detection is taking place.
01:10.19WIMPyThank you very much.
01:10.45wizard171Sorry for the delay ... got sidetracked on my end ... glad its working !
01:11.17WIMPyThat's fast enough for me :-)
01:13.58WIMPyNow I can try to find ouy why it is sending a redirecting number.
01:15.33WIMPyBTW: Is tehre any reason, why it defaults to
01:15.36WIMPywct4xxp 0000:03:08.0: VPM450: hardware DTMF disabled.
01:25.17wizard171ah, 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 ... :)
01:25.48WIMPymodinfo helps :-)
01:26.01WIMPyI just wondered if there was a reason.
01:27.10wizard171None I have come across ... other than the "asterisk is supposed to do it" kind of thinking that was built into it in the beginning ...
01:27.43WIMPySounds like I want it on then.
01:28.35wizard171ah, 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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01:29.37orionihi there, i want to analyse the acd/asr for the cdr on mysql
01:29.52orioniis there any ready script/application i can try ?
01:29.55WIMPyYou think the algorithm running on the DSP is not as good as that in Asterisk?
01:31.12*** join/#asterisk justdave (~dave@unaffiliated/justdave)
01:32.22wizard171What, 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?
01:33.57wizard171Ergo, 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 ... :)
01:34.50WIMPyI'll try it out.
01:35.47wizard171Well ... I hope ... all your buttons get pressed ... just the way you want ... eh ?   :)
01:35.59orioniis there anyone that does reporting on the cdr`s on mysql for acd/asr ?
01:36.26orionii would like some example since my queries are executed after 11 seconds and i want sth faster than that
01:36.40WIMPyIf it can do THAT ...
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02:34.05dandate2how do you do a conference call with an ATA device? i.e. sip 1 is talking to sip2, sip1 adds sip3 to the conversation
02:34.22dandate2its so easy with xlite =@
02:35.17WIMPyTransfer bot to a MeetMe() or ConfBridge() extension.
02:35.22WIMPyboth
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02:38.02dandate2this is soemthing i have to configure at the * end or the ATA end?
02:38.11dandate2i would have assumed the ATA end since xlite has it built in
02:40.40dandate2this wouldnt be practical for us since typically sip2 is on a cell phone and mentally challenged
02:40.52dandate2when using * conference, each person has to dial a code huh
02:43.06dandate2so it is the ata device that does the 3 way calling, god damn this thing for not having a manual
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02:51.15dandate2nm i guess we just had to use the flash button heh
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02:52.42matt_vulooks around
02:55.23wizard171matt_vu: ... you are ... it ...
02:56.28matt_vuasterisk is a pbx?
02:56.45wizard171only if you configure it that way ... ;)
02:57.27wizard171I have no luck ... configuring it to bring me women ... :)
02:57.44matt_vui see, so all the discussions here are about 'as a SIP'?
02:58.01matt_vu:)@wizard... try harder
02:59.36wizard171ah, SIP ... the coffee, other times we IAX2, skinny, gtalk, h323, etc. ... some even ... dahdi ...
03:01.26matt_vudo u mean gtalk is on asterisk?
03:03.31wizard171yes, you can gtalk on asterisk ...
03:04.03matt_vuic
03:05.56wizard171I guess you could say ... you can ... jabber while you jingle on your gtalk ... ;)
03:06.10matt_vu:)
03:15.55ldiamondIs there a setting to have sip poke retries when theres no response?
03:17.16ldiamondI'm talking about the peer poke here
03:18.27wizard171Ah, if you set "qualify=xxx" where xxx = some numeric value, that is how often it "pokes" ... is that where you are going ?
03:19.14wizard171If you have it set "qualify=yes" ...  you get 2000ms (I think, as default) ...
03:19.17WIMPyqualify is the maximum time to wait for an answer.
03:19.25WIMPyyes
03:19.51cmendes0101Is there a way to pass variable using originate command in CLI?
03:20.11wizard171It's "how often to check" in "peer" terms, I believe ...
03:20.31WIMPyNo, that's qualifyfreq.
03:21.17WIMPyAnd I think thare is a qualifyfailfreq or something as well. Don't see that ATM.
03:22.48WIMPyHmm. Maybe that was IAX only.
03:23.40wizard171You 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" ...
03:24.09WIMPyNo. Looks like that is indeed iax only.
03:26.03wizard171cmendes0101: what do you have in mind?
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03:26.39wizard171s/mind?/mind? (source vs. value)/
03:27.16WIMPyBut sip does have different values internally as well. I just can't find a way to configure it.
03:29.38wizard171well ... the source code, usually does not start ... telling stories ... until after you compile it and run it ... eh ? :)
03:30.05cmendes0101wizard171: 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?
03:30.26ldiamondLooking at this: https://issues.asterisk.org/bug_view_page.php?bug_id=16936
03:30.26WIMPys/re it/re it at runtime/
03:30.40ldiamondMaybe this might help fix my issue
03:30.50ldiamondupgrading to newer version.
03:30.56cmnkyMillions affected as Skype goes down
03:30.56cmnkyWed Dec 22, 6:04 pm ET
03:33.38wizard171cmendes0101: that originate is something a human is typing ? (if not, can the script substitute what you want?)
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03:35.17wizard171ldiamond: what version are you running?
03:35.23cmendes0101wizard171: 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
03:35.45ldiamond1.6.2-5 or something
03:35.57ldiamondI am updating it now.
03:41.39wizard171cmendes0101: ah, instead of "s", can the script specify "this1" or "that1", which can be defined as "exten => this1,blah" , etc. ?
03:42.31wizard171it would mean many different versions ... or, creative dialplan ... :)
03:42.49ldiamond1.6.2.5-0ubuntu1.1
03:44.16wizard171ldiamond: the "fix" did not make it until 1.6.2.8 (according to my changelog) ...
03:45.23wizard171or should I have said the "fix of the fix, which was fixed ...", etc.
03:45.48cmendes0101wizard171: 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
03:46.47ldiamondarghhh, and ubuntu doesnt update fast enough
03:46.58wizard171cmendes0101: well ... "its a command line" ... it can "vary" by whatever your script feeds it ... :)
03:47.37wizard171ldiamond: ah, but ubuntu does ... much else ... very well ... eh?
03:48.17ldiamondwizard171: 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
03:49.28wizard171ldiamond: (did I just have "dejavu"? ala "The Matrix") ... Goto the Source my friend ... it only starts ... telling stories, after you compile it ... :)
03:49.43cmendes0101wizard171: actually if I do something like exten => _XXXX, 1, Meetme(${exten}) that should work right?
03:49.48cmendes0101well I guess ill just try it lol
03:50.05p3nguinpbx_config won't like spaces.
03:50.51wizard171cmendes0101: yeah, and you should "filter" that "${exten}" for "valid" as well ... ;)
03:51.22ldiamondwizard171: 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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03:54.28wizard171ldiamond: 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 ...
03:55.08wizard171ldiamond: just don't "make samples" ... eh?
03:55.38ldiamondI'm scared about sudo apt-get remove asterisk.... my config will most likely be gone..
03:57.24p3nguinlrn2cp or lrn2rsync
03:57.39wizard171ldiamond: "host# cp -r /etc/asterisk  /path/to/keepit" ... same/same for "dahdi", etc.
03:58.06ldiamondyea, it's the 'etc' part there that scares me :p
03:58.58wizard171ldiamond: (sigh) ... yeah, that's the one I usually forget ... :)
03:59.29wizard171lf you have another machine to practice with ... that would be good ...
03:59.34p3nguinReally, I wouldn't expect apt-get to delete your files unless you use purge.
03:59.50p3nguinI'd still save my backups, though.
04:00.00p3nguinIt only takes a second to make a backup.
04:00.05p3nguinliterally
04:00.22WIMPywats that machine!
04:00.40wizard171or, you could "image" the drive onto another ...
04:01.07WIMPyMaybe use RAID? *cough*
04:01.52p3nguinIf you delete a file on RAID, it gets deleted everywhere.
04:02.13wizard171just much faster ... than mirror ... or JBOD ...
04:02.58p3nguinIt takes only a fraction of a second to clear an inode from being used.
04:04.54wizard171ah, but, compiling from the source = much better ... IMHO ...
04:05.13p3nguinThe packages were built using the same source.
04:06.07wizard171meaning, worth the effort, to get what you want/need ... and sometimes, optimized for what you have ... eh ?
04:07.27p3nguinUnless you're running 25-year-old hardware, you're not likely to see any performance gains by "optimizing" your own compile.
04:07.53p3nguinThe packages provide the exact same software that you'll get if you compile it yourself.
04:08.31WIMPyEither the package won't even run un 15 year old hardware or you have a good chance of about 10%.
04:09.53wizard171I 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 ...
04:11.28p3nguinAnd 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.
04:11.50p3nguinIf you think I'm wrong, benchmark it and show me the results.
04:14.08wizard171hmm ... 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 ...
04:18.13p3nguinI'd have a fair amount of faith in the guys who maintain the packages every day/week/month/year for a given distro.
04:19.32p3nguinNot that it really makes any difference for me, since I build my own packages, but I think I made my point.
04:20.14wizard171well ...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" ...
04:26.18wizard171or, 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 ... :)
04:27.15wizard171makes me feel ... special ... ;)
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04:36.46brainiacDoes 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
04:36.49brainiac?
04:39.41WIMPy"dialplan reload"
04:42.51wizard171brainiac: does "add the DID" mean changes to "chan_dahdi.conf"? (very little in there can be changed without a "restart" at least) ...
04:44.32WIMPyYes, took me quite some time yesterday to remember that restarting Asterisk isn't enough when reconfiguring interfaces.
04:45.17p3nguinYou ought to be able to reload the necessary module(s) and get changes to take effect, I'd think.
04:45.41brainiacI'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
04:45.56p3nguinI don't know; I just don't reboot my computers.  It irritates me, for some reason.
04:46.03WIMPyYes, I had to rmmod and modprobe. But I'm not used to that.
04:46.08wizard171And you thought all that ... "modprobe"(ing) ... was just for ... fun ...
04:47.58wizard171brainiac: 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?)
04:48.53brainiacinbound calls are going to a particular extension depending on the number dialed.
04:53.24wizard171well, 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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04:58.16brainiaccontext=TE110P/4 (zapata.conf not chan_dahdi.conf)
04:59.39brainiacThis is only a 1 channel card (TE110P).  I think this should say context=TE110P/1. Right?
05:00.08WIMPyYou can call it whatever you like.
05:00.13wizard171brainiac: 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?
05:00.40wizard171in "extensions.conf" or "extensions.ael" ?
05:01.32brainiacit lives in globals
05:01.40brainiacextensions.conf
05:01.41p3nguinFail.
05:03.25wizard171Okay ... in "extensions.conf" ... you REALLY need some other "[context-name]" style entries ... or an "#include ..." that has them ?
05:03.55brainiacok
05:04.11p3nguinYou need [general], [globals], and THEN contexts where you put useful extensions.
05:04.31wizard171where the "[context-name]" part has some "exten => ... " type entries ... underneath it ...
05:04.39p3nguinI'd even add a [default] in there after [globals] just to be thorough.
05:07.32wizard171Once 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 ...
05:07.44brainiacok
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05:07.50p3nguin~book
05:07.50infobot[~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/
05:07.54p3nguinI guess no one reads anymore.
05:08.16p3nguinToo much work, perhaps.
05:08.29azlonwhere can i find an asterisk ubuntu 10.10 repo?
05:09.36drmessanoDownload and install from source.. you don't want a repo
05:09.44azlonhrmm
05:09.46azlonok
05:09.54azloni will search for a guide to install from source
05:09.55azlonthanks
05:15.05brainiacthanks 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.
05:15.28azlonhehehehehe
05:15.35azlonnae!
05:15.39azlonnow click on living room
05:15.43WIMPyToo many books.
05:15.49azloni set ioops
05:15.52azlonwrong room
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05:18.12wizard171Yeah, I gave up "reading" and started "experimenting" ... and there are those (unkind souls) that say, my "experiments" never should have made it into "production" ... :)
05:18.59wizard171Just what I get for creating ... "art" ... eh?
05:19.52wizard171or, maybe they think I left the "f" out of that word ... ?  (who knows?)
05:22.38WIMPyThat's what I heard until I remembered to reload the kernel drivers.
05:23.14wizard171Yeah ... those kernels ... will drive you, alright ... LOL
05:24.16wizard171I am SURE ... "popcorn" ... had nothing to do with it ... eh?
05:25.14WIMPyIt wasn't me. It was my dahdi.
05:25.37wizard171I know ... it just wants to ... zapata ... ?
05:28.19wizard171I am going to ... go sleep ... before this devolves into ... "extensions" ... ;) I have enjoyed, until we meet again ... enjoy!
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06:52.43drmessanoO.o
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07:27.37schmidtsgood morning
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07:32.28drmessanogm
07:33.40schmidtseverybody allready in christmas mood?
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07:50.47Russanyone deal with support issues with Level 3?
07:55.12drmessanoschmidts, I hate christmas, so no
07:58.03schmidtsdrmessano i can understand this :D
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08:56.20arekmhi, I'm in "s" context. Can I somehow figure out what extension was dialled anyway?
08:57.47schmidtsarekm 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)
08:59.27Corydon76-homeUnless he's using a Macro, in which case, the EXTEN will be in ${MACRO_EXTEN}
09:11.00joobiehey gusy
09:11.48*** join/#asterisk devyll (~paul@thpallady.net.hostway.ro)
09:14.01devyllhello 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
09:14.02devylln'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)
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09:39.15kaldemardevyll: why not use context, extension and priority in the originate?
09:40.03kaldemarand use an extension that lets you have a part of the extension as the argument.
10:03.53devyllkaldemar, yeah .. it would work.. resolved with call files though
10:03.54devyllthanks
10:43.38*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
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10:53.32X-Raimoquestion about realtime. How asterisk searches for number: local and then realtime or vice versa?
10:53.53schmidtsX-raimo with number you mean extension?
10:54.03*** join/#asterisk sekil (~sekil@80.93.247.26)
10:57.00X-Raimoschmidts: yes
10:57.59schmidtsx-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
11:01.52X-Raimoschmidts: 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.
11:02.39X-Raimoshould we define srtpcapable=yes and encryption=yes for 1999 in sip.conf?
11:05.41schmidtsX-Raimo you are mixing extensions and sip peers/users
11:06.13schmidtsor do you have a peer named 1999 which is a physical phone or server which do the echo test?
11:06.45X-Raimoschmidts: server's echo test
11:08.07schmidtsX-raimo is it exten => 1999,1,Echo() or exten => 1999,1,Dial(SIP/serverforecho) ?
11:09.06X-Raimoschmidts: exten => 1999,1,Echo()
11:09.24*** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com)
11:09.25schmidtsok, sorry but i just want to be sure ;)
11:10.39schmidtsit 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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11:21.43X-RaimoIs phone connected via SRTP uses same codecs as phone connected without SRTP?
11:22.36b14ckX-Raimo: yes.
11:23.51*** join/#asterisk tasca (~tasca@189.4.108.113)
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11:25.57marksaitishey. does anyone know which package /usr/local/ssl came from in centos5?
11:38.15Tim_Toadymarksaitis try yum whatrovides
11:39.12Tim_Toadybut since its in local it must be from some compiled by hand source package
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12:11.02azlonis there a gui for the new asterisk 1.8?
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12:45.36X-RaimoIs option encryption in sip.conf has only 2 values: "yes" and "no"? What other options does it have?
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13:02.48*** part/#asterisk azlon (~ryan@94.29.137.166)
13:04.08anonymouz6661.4.38 is the latest bug fix release, right? 1.4.39 and so on will be released only if there's security fix
13:04.47leifmadsenanonymouz666: no, 1.4.39-rc1 is out and will become the next release if there is not an rc2
13:04.54leifmadsenif there was a security release, it would be tagged at 1.4.38.1
13:05.16leifmadsenonly in the 1.6.x series do we increase the numbers on a security release (since 1.6.2.15.1 looks silly)
13:05.45leifmadsena 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)
13:06.04anonymouz6661.6.2.15.1-rc3 looks fine
13:06.06anonymouz666hehe
13:06.17leifmadsenthe 1.4 and 1.6.2 series have had their bug fix statuses extended to April 2011
13:06.26leifmadsenwe would never have a 1.6.2.15.1-rc3
13:06.35leifmadsenthe .1 on the end would be released directly without an RCs
13:06.50leifmadsenbecause it would be either a bad regression, or a security release which would not warrant a release candidate
13:06.55anonymouz666ahh it was extented then, I remeber reading the latest 1.4 release on 21 december 2010.
13:07.02leifmadsenyes, that was before astricon :)
13:07.18leifmadsenwe extended it at astricon so people would have more time to migrate/test 1.8
13:07.45leifmadsenanonymouz666: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
13:08.34leifmadsen~asterisk-versions
13:08.34infobotit 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
13:10.02leifmadseninfobot: 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
13:10.02infobotokay, leifmadsen
13:10.09leifmadsen~asterisk-versions
13:10.09infobotInformation 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
13:10.37leifmadsen~coffee
13:10.37infobotfrom 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
13:10.42russellbleifmadsen: back to work
13:11.00leifmadsenrussellb: I'm tired of writing about queues :)
13:11.09russellbleifmadsen: write about DUNDi then
13:11.13leifmadsenbah!
13:11.22leifmadsenthis book should write itself
13:11.34leifmadsenI just want my shiny nickle!
13:11.40russellbheh
13:11.57leifmadsenqueues is already up to 32 pages...
13:11.58russellba whole nickel?!
13:12.12russellbI was only promised a pretty penny.
13:12.16leifmadsenya, but it'll be a US nickel which is worth less than 5c
13:12.25leifmadsenrussellb: you get upgraded to nickel on the third book
13:12.54russellbgoes back to work :-p
13:12.56schmidtsyou guys think you get paid for this? i though you have to pay to be allowed :D
13:13.07russellbpretty much
13:13.34schmidts:D
13:13.34leifmadsenjoinempty! ooo!
13:14.13drmessanoHows that entire chapter on Answering Machine Detection coming along?
13:14.21drmessanoThat 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
13:17.51coppicea whole chapter of evil
13:18.23leifmadsendrmessano: we're not even bothering :)
13:18.32leifmadsenmaybe in the 4th edition...
13:19.04drmessanoor in the red leather bound 666th edition?  MUHAHAHAHA
13:19.43fileleifmadsen, less complaining more writing or the electrocution will continue!
13:19.44leifmadsenyes
13:19.54leifmadsenfile: and you're not kidding
13:20.20coppicethere's nothing wrong with the ethics of AMD that cash can't cure
13:20.35leifmadsen:)
13:25.11*** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106)
13:25.38*** join/#asterisk atan (~atan@unaffiliated/atan)
13:28.01coppicedoes anyone do a usable AMD for *?
13:28.48russellbnot that I know of
13:29.06coppicesurely people like vicidial must have something
13:29.25russellbtrue ... I think I just pretend that evil like that doesn't exist
13:30.19anonymouz666hah
13:30.24anonymouz666poor vicidial
13:30.33*** join/#asterisk wizard171 (~wizard@h47.50.20.98.dynamic.ip.windstream.net)
13:31.12SuPrSluGI tried using AMD. 80% is about the best you can hope for.
13:31.32russellbI've heard some optimistic accounts of 90% ...
13:31.55coppiceyou mean the app that is in the * distribution? that one is too mickey mouse to be of any real value
13:31.57SuPrSluGthat's pretty good.
13:32.15russellbAsterisk: The Mickey Mouse PBX
13:32.26*** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn110.78-98-245.t-com.sk)
13:33.17coppiceshhhh. don't let disney hear you say that
13:33.41russellbeep
13:40.05russellbMickey Mouse is a registered trademark of Disney. Asterisk is not endorsed by Disney (though they probably use it).  (Please don't sue me.)
13:40.31wizard171Nah, that would be ... Goofy ... :)
13:40.51*** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn111.78-98-213.t-com.sk)
13:41.22pathis it possible to detect when someone hangs up?
13:41.38pathI need to use GotoIf and some Channel status
13:42.03pathi.e. dial some extension if someone hangs up
13:47.54*** join/#asterisk grkblood13 (967da669@gateway/web/freenode/ip.150.125.166.105)
13:48.35grkblood13i 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?
13:49.15marksaitisin sip.conf, how do I configure a user to have a password? i.e. user [101]
13:49.54SuPrSluGsecret=
13:49.59wizard171marksaitis: its a "secret=xxx" ... shh don't tell ...
13:50.12*** join/#asterisk slackytude (~slacky@drms-590df9cf.pool.mediaWays.net)
13:53.02marksaitisok
13:54.39schmidtsmarksaitis but you should not use users like 101 or something, cause its not secure
13:56.25pathhow can I dial to some extension if somebody hangs up before any dtmf?
13:58.14wizard171path: sorry, I do not understand ... you mean, once the phone goes "off-hook"?
13:59.02paththe 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)
13:59.58pathI need to be able to log that anyway, so I want to grab that info
14:00.39marksaitishow to set CLI verbosity to its max? I get file open fail error, wanna know more info
14:00.54pathcore set verbose 50
14:00.57pathcore set debug 50
14:01.06marksaitisthanx
14:01.28X-Raimois 50 maximum debug value?
14:01.40marksaitissame error still :))) whats the point in this verbosity thing
14:03.04wizard171path: I presume you are using "Dial" ... and either "M(acro)" or "G(osub)" ... and in there, using "Read" to capture what they press?
14:03.34pathyes
14:03.46pathbut I forgot cdr exists.. so
14:04.26wizard171and perhaps ${DIALSTATUS}
14:04.30pathI tried it
14:05.04pathe.g. exten => s,1,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?s,12)
14:06.18wizard171marksaitis: the point to verbosity ... sometimes you get ... more words ... for the same error ... eh?
14:06.25marksaitisanybody:
14:06.26marksaitis<PROTECTED>
14:06.58marksaitisgoodness sake, I already gave chmod 777 and chown asterisk to all these cert files, every single one :)
14:07.03marksaitisstill no go
14:07.10X-Raimomarksaitis: I had this problem yesterday.
14:07.18marksaitiswhat did u do?
14:07.19marksaitis:)
14:07.38X-Raimomarksaitis: I solved it by changing my client's softphone
14:07.54X-Raimophoner lite works well.
14:08.00marksaitisso u recon its softphone
14:08.05marksaitisphoner lite? tls srtp?
14:08.20X-Raimomarksaitis: yes,sir
14:08.23marksaitisdamn softphones, whats wrong with these f*ckers
14:08.47X-Raimoportgo gives an error
14:08.56marksaitisI just tried bria on iphone
14:09.16X-Raimonow I have problems with srtp
14:09.17marksaitiswhy the hell does it not say what file....
14:09.18marksaitis;]
14:09.31X-Raimostill dunno
14:10.07X-Raimothere is *-file exists. It's actually an error message
14:10.37marksaitisI am totally getting pissed of by this tls cosmos
14:10.39marksaitis;]]]
14:10.47Kattymorning
14:14.04*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
14:14.11marksaitisdear oh dear
14:16.37Kattyhello my asterisk does not work at all how to fix pls
14:17.12seanbrightKatty: #asterisk
14:17.15seanbrightoh wait...
14:17.25X-RaimoKatty: what error does it gives?
14:17.37KattyX-Raimo: what is error
14:17.39marksaitisJust tried from other softphone
14:17.40marksaitis[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
14:17.40KattyX-Raimo: it does not work
14:18.26X-Raimomarksaitis: transport=tls
14:18.57X-RaimoKatty: give us /var/log/asterisk/messages
14:19.00marksaitisX-Raimo ?
14:19.09marksaitisI have transport=tls
14:19.10Kattygiggles hysterically
14:19.18Kattyaww you thought i was serious
14:19.21Kattythat's /so/ cute
14:19.39X-Raimomarksaitis: you should enter this in sip.conf
14:19.48Kattypats X-Raimo
14:20.04X-Raimoin peer's section
14:20.46marksaitishttp://pastebin.com/PjZDKPei
14:20.50marksaitisplease have a look
14:24.25X-Raimomarksaitis: there is what I have: http://pastebin.com/RG3jQHGW
14:24.44X-Raimomaybe it gonna be useful for you
14:25.13marksaitisok. thanks for sharing
14:27.03*** join/#asterisk azlon (~ryan@94.29.137.166)
14:27.15azlonanybody here use PBX in a Flash?
14:28.11luckman212azlon: yes I do
14:28.48azlonluckman212: i just installed PIAF and for some reason i cant log in on the webmin
14:29.02azlonluckman212: everything i have read says that my user/pass should be the same as my linux login
14:29.27azlonluckman212: i ran passwd-master so they would all be the same, but it doesnt work... keeps giving me the login prompt over and over
14:29.54azlonluckman212: am i doing something wrong?
14:30.04luckman212have you tried running "passwd-maint" from the SSH console?
14:30.23azlonno
14:30.28azlonis that different from passwd-master?
14:31.06azlontrying it now
14:31.07luckman212hmm, well normally you would use the maint pwd to log into the web gui
14:31.21luckman212passwd-master should work also
14:31.35luckman212which piaf version did you install?
14:31.40azlonuhmm
14:31.46azlonthe newest one... let me find the version
14:32.10azlon17554
14:32.19*** join/#asterisk Vrtigo1 (~Vrtigo1@vpn.lpga.com)
14:33.16azlonsweet!
14:33.18azlonit worked!
14:33.19azlonthanks!
14:33.24marksaitiscan anybody tell me PLEASE what does this mean:
14:33.24marksaitis[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
14:33.29marksaitisPLEASSSSSSEEEEEEE
14:33.33Vrtigo1anyone know why the record app has a 20 minute limit in 1.6.2.2?
14:33.47luckman212azlon: cool, glad to hear!
14:33.50*** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2)
14:34.02luckman212enjoy your Pbx in a flash... its a lot of fun to play with
14:34.57azlonluckman212: do you know where i can get info on how to setup a GV account?
14:35.07azlonyeah, this thing is pretty impressive so far
14:35.21azloni am in kuwait right now and want to setup a system so my guys can call back to the states easily
14:36.17luckman212GV support is built into the version you have, just run the incredible pbx installer and it walks you thru...
14:36.33marksaitisI did a shit on this fucking tls
14:37.08azlonhrmm
14:37.15azlonok
14:37.36luckman212azlon: http://pastebin.com/kPLPx3S0
14:37.50azlonthanks!
14:39.31*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
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14:40.52*** part/#asterisk vasanthv16 (~Nithya@216.195.17.249)
14:50.09azlonluckman212: is incredible PBX an addon to PIAF or are they two different things?
14:50.28luckman212it's an addon for PIAF
14:50.33azlonok
14:51.13luckman212gives you a bunch of extra features.  read up: http://nerdvittles.com/?p=712
14:51.32*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
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14:55.53frigidzephyrWIMPy: did you ever resolve teh "not available from userspace" issue with the TE220 firmware?
14:56.04frigidzephyrif not, you might consider calling Digium technical support to report the issue
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15:24.17Kattystretches
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15:41.07*** join/#asterisk jbesclapez (~jbesclape@41.208.164.78)
15:41.33jbesclapezgood morning or evening depending on where you are....
15:41.46hurdmanhow use dialplan function : round and/or floor into an 1.6.* ? i can't found anything
15:41.48jbesclapezIt is my first step in this channel...
15:42.58p3nguin~ask
15:42.58infobotQuestions 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.
15:43.02p3nguin~answers
15:43.02infobot[~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
15:43.09p3nguinjbesclapez: ^^^^^^^^^^
15:43.16Kattyomnomnoms snack
15:43.43jbesclapezI 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.
15:44.16p3nguinIf you think you know how to use Asterisk because you've used Trixbox, you may very well be wrong.
15:44.35schmidtsi wish you all merry christmas!
15:44.50jbesclapezprobably p3nguin you are right... i was just pointing that to tell you that i am not a total newbie
15:45.00jbesclapezschmidts: thanks and to you too. :-)
15:46.13jbesclapezHowever, 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.
15:47.12p3nguinYou'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)
15:48.12jbesclapezp3nguin: what would you like me to proove you exactly?
15:48.17p3nguinAsterisk 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
15:48.42p3nguinjbesclapez: A sip debug would be a great place to start.
15:48.48_Corey_p3nguin: nope
15:48.59jbesclapezOK...
15:49.08_Corey_I've seen this a few times before on other systems but have never been able to explain it
15:49.14p3nguin_corey_: Nope it doesn't, or nope I'm not right?
15:49.34jbesclapezlet me google how to do a sip debug :-)
15:49.34_Corey_p3nguin: no log rotation is configured in logger.cnf
15:49.48jbesclapezyou 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
15:52.07p3nguinjbesclapez: 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
15:52.50p3nguinjbesclapez: 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
15:53.11jbesclapezhow do i send a sip debug? where to look for that info? in the call detail records - this one?
15:53.43p3nguinAnd then pastebin the entire debug output after you collect it.
15:53.43leifmadsen_Corey_: sounds like your logger.conf is setup to log lots of information
15:54.02jbesclapezi am googling this ... :-)
15:54.05_Corey_leifmadsen: yeah: full => notice,warning,error,verbose,debug
15:54.06leifmadsen_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.
15:54.10leifmadsenwell there you go
15:54.14leifmadsenthat's why it fills up
15:54.15*** join/#asterisk WonTu (~WonTu@p57B53D77.dip.t-dialin.net)
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
15:55.21p3nguinjbesclapez: There's nothing to google.  I gave you exact and precise instructions.
15:55.27leifmadsenshrugs
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
15:56.03jbesclapezthen i must be stupid because i dont know what you mean or where to find the sip debug....
15:56.17p3nguinBut 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
15:56.39p3nguinjbesclapez:   <p3nguin> jbesclapez: Depending on your asterisk version, you enable sip debug from the CLI by "sip set debug"
15:56.42p3nguin<PROTECTED>
15:57.02p3nguinNote: someone who knows how to use asterisk would know this.
15:57.11p3nguinJust sayin'.
15:57.15_Corey_p3nguin: They Asterisk CLI is saying "Asterisk Event Logger restarted" and "Rotated Logs Per SIGXFSZ (Exceeded file size limit)"
15:57.18leifmadsenor read the documentation :)
15:57.23*** join/#asterisk JonnyD_work (~Jon@173.226.80.154)
15:57.31makafregood 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
15:57.51leifmadsenmakafre: you need lib-iksemel and whatever other deps
15:57.57leifmadsenmakafre: http://ofps.oreilly.com
15:58.16leifmadsenmakafre: don't install via a patch, because it's already part of asterisk
15:58.18p3nguinI no longer expect people to be how to read documentation.  All your hard work of writing docs is in vain.
15:58.33makafre@leifmadsen: ah, serious, I thought I had to merge it
15:58.35leifmadsenmakafre: notice how that review is closed as "submitted"
15:58.46leifmadsenthat means it's been submitted and merged into the project already
15:58.49leifmadsenp3nguin: amen
15:59.14makafre@leifmadsen:  I see....and how do I check in which release it was merged initially?
15:59.22p3nguins/be how/know how/
15:59.34p3nguinWhen did Asterisk get built-in log rotation?  I use 1.4 and I have to let my system rotate the logs.
16:00.05leifmadsenmakafre: 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/
16:00.29leifmadsenmakafre: then scroll near the bottom and look at the message that says what revision it was merged in
16:00.44leifmadsenSeptember 25, 2009
16:01.24_Corey_This customer's box is actually running 1.2.14
16:02.13jbesclapezp3nguin: 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
16:04.04p3nguinjbesclapez: Magic, maybe?  Most people believe that's how asterisk works.
16:04.23tvc123p3nguin: most people believe that about computers in general
16:04.29p3nguins/asterisk works/computers work/
16:04.36p3nguinbetter?
16:04.45jbesclapezyea because most of the people are stupid right ? ;-)
16:05.03p3nguinA lot of people are pretty stupid, but I don't know if most people are.
16:05.14tvc123jbesclapez: their not stupid until they piss me off ;)
16:05.49makafre@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?
16:05.54jbesclapeza 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
16:06.25p3nguinThat's kind of retarded...
16:06.27jbesclapezshe is using a mac probably no?
16:06.34leifmadsenmakafre: 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
16:06.35Kattypeeks in
16:06.43_Corey_no, they're not available in pink from apple i don't think
16:06.48leifmadsenmakafre: again.... ---> http://ofps.oreilly.com
16:06.52_Corey_lol
16:06.53p3nguinIt's pink, so everyone should know you have to get rough with it and sometimes smack it around a bit.
16:06.53leifmadsenit's all documented there
16:06.56jbesclapezlol
16:07.12_Corey_I like to feed people tech psychosis
16:07.19makafre@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
16:07.32leifmadsen_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)
16:08.46jbesclapezwhat is a roomba (sorry)
16:08.55leifmadsenrobot vacuum
16:08.57_Corey_the robotic vacuum
16:09.23jbesclapezeh google told me ;-)
16:10.11leifmadsenya, you should have just done that first :)
16:12.05jbesclapezyep 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
16:14.23jbesclapezOK everybody.... have a nice christmas and thanks especially to p3nguin
16:14.42jbesclapezbye
16:15.18*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
16:15.55*** join/#asterisk coppice (~chatzilla@210.17.219.137)
16:30.46Kattywtb vicks vapor rub
16:31.51*** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452)
16:35.21makafre@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
16:35.38leifmadsenit's possible JABBER_RECEIVE is only on 1.8 then
16:35.46leifmadsendid you look in res/funcs for it?
16:35.59makafrelet me verify
16:36.38*** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de)
16:36.53makafreyep, it's not there
16:37.12leifmadsengrep -r "JABBER_RECEIVE" *
16:37.46leifmadsencould probably look in CHANGES to see if it was added
16:37.50leifmadsenit should list it if it was
16:37.52makafreit returns nothing, its definetely not there
16:38.02leifmadsenthen it's a 1.8 only feature
16:38.15makafreokido
16:38.57makafrethen I am stucked  :- ) lol
16:39.10azlonluckman212: when i dial my GV number, nothing happens... i ran the incredible pbx like you said and entered my user/pass
16:40.03luckman212azlon: what do you see in your * console
16:40.50azlonluckman212: nothing
16:41.03*** join/#asterisk Tim_Toady (~moi@77.49.3.151.dsl.dyn.forthnet.gr)
16:41.03azlonit is ringing right now and it just says "pbx*CLI>"
16:41.12luckman212core set verbose 5
16:41.32*** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net)
16:41.40luckman212type that & try again
16:41.41azloncall it again now?
16:41.43azlonok
16:41.58luckman212are you trying inbound --> your pbx or outbound?
16:42.06azloninbound
16:42.30luckman212have you set up your NAT/firewall properly?
16:43.14azlonuhmm
16:43.18azlonprobably not...
16:43.28azloni dont have a firewall yet
16:43.58azlonim trying to figure out how to copy from this VM
16:44.03azloni need to ssh into it
16:44.25luckman212you're running your pbx in a vm?
16:44.39azlonluckman212: just until i figure out how it works
16:44.43azlonthis is just temporary
16:44.48luckman212where is this vm? behind your nat?
16:44.58azlonyes
16:46.01luckman212does 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?
16:46.20azlonit is bridged
16:46.29azlonit is VMWare 7
16:46.56azlonok
16:47.01azloni ssh'd into it
16:47.05azlonthis is what it is giving me
16:47.06azlonhttp://pastebin.com/ci5LunrC
16:47.26luckman212that looks ok, now try to make an inbound call
16:48.13azlonmore of the same thing
16:48.20azlonit just keeps posting that same message over and over
16:48.43leifmadsenya that msg isn't related to GV
16:48.44azlonlet me try calling from a different number
16:48.53luckman212your inbound call is not even hitting the pbx
16:48.58luckman212you've got a NAT/firewall issue
16:50.03luckman212have you followed the instructions?
16:50.19luckman212you need to fwd UDP 5222 to your pbx for GV inbound
16:50.47azlonno, i didnt see any instructions
16:50.53azloni just followed the pastebin you gave me
16:50.59azlonthen followed the onscreen instructions from there
16:51.16luckman212I pasted in the link to the full instructions. maybe you missed it
16:51.17Nuggethttp://bash.org/?35339
16:51.21luckman212please read it
16:51.41azlonok, brb
16:51.49*** join/#asterisk brah (~brah@host92.200-82-54.telecom.net.ar)
16:52.02*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
16:52.25azlonerm... what am i looking at on there?
16:52.33azlonthis 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
16:59.46luckman212try here.... http://i.imgur.com/jDhrt.png
17:01.13azloncool, still reading through the first one
17:01.23azlonthis thing is freaking crazy... it does EVERYTHING!
17:03.54luckman212easy now, don't get ahead of yourself
17:04.13luckman212:D
17:11.47QwellPlease tell me you aren't actually trying to use the PBIAF howto for gV..
17:12.46Qwellhttps://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
17:26.37azlonhrmm
17:26.39azloninteresting
17:26.44azloni think im burned out for the night
17:27.03azlonthanks for your help, luckman212. ill pick it up again tomorrow
17:27.44luckman212sure... come back & let us know how you get on
17:27.46*** join/#asterisk timahvo1 (~rogue@41.223.57.74)
17:30.56*** join/#asterisk Defraz (~Defraz@63-226-95-147.dia.static.qwest.net)
17:33.23*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
17:35.07anonymouz666shrimp
17:39.56*** join/#asterisk wizard171 (~wizard171@h25.199.91.75.dynamic.ip.windstream.net)
17:41.44makafre@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...
17:42.05leifmadsenNo, I never recommend backporting something unless you have the skills to do so
17:42.14makafreok
17:42.24makafresure
17:43.51Qwellbackports leifmadsen to Asterisk 1.2
17:43.57leifmadsenoh god no
17:43.58Qwelldeprecates the backported leifmadsen.
17:44.01QwellEOL
17:44.47coppiceleifmadsen: so, in general you wouldn't recommend doing something you can't?
17:44.58leifmadsencoppice: precisely :)
17:47.19paulcbacks away nervously
17:48.08paulcPlaying 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)
17:48.45leifmadsenpaulc: 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)
17:52.41paulc@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") ;-)
17:53.05leifmadsenpaulc: ya, Monitor() is good when you want separate files for each channel
17:53.55paulcand 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
17:54.56leifmadsenpaulc: 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
17:55.17leifmadsenwell StopMonitor() is required to stop recording, but it'll stop upon hangup too
17:55.49leifmadsenok I'm off to start slow cooking a roast for dinner
17:56.18paulchangup 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?
17:56.26paulcMmm roast dinner.. sign me up! :)
17:57.08leifmadsenpaulc: that's where it starts getting complicated (and don't even both thinking about transfers), so you'll have to experiment
17:57.30paulcLOL fair enough.. maybe the force be with me! :)
17:57.30leifmadsenremember that when you transfer the monitoring follows the channel where Monitor() (or MixMonitor()) were triggered from
17:57.54leifmadsenso 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
17:58.07leifmadsenA (Monitor) --> B --> transfer A --> C (call is recorded)
17:58.25paulcTransferring 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)
17:58.26leifmadsenA (Monitor) --> B --> transfer B --> C (A hangs up, call stops being recorded)
17:59.02Kattyhttp://i.imgur.com/ZfGQX.jpg <- ^_________^
17:59.18paulcShouldn'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
17:59.46paulcKatty: I don't get it? (still feeling fuzzy - it's early!)
17:59.51leifmadsenpaulc: that sounds easier :)
18:00.04Kattypaulc: that is molecular structure of glucose
18:00.05paulcKatty: Ah.. chemical reference?
18:00.05leifmadsenya I don't get it either
18:00.10leifmadsenoh! :)
18:00.19anonymouz666leifmadsen: 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)
18:00.20KattySugar Cube
18:00.22paulcTalk about geeky number plates :)
18:00.30Kattyi'd totally drive it
18:00.38leifmadsenanonymouz666: ya now you're getting into voodoo :)
18:00.46leifmadsendetecting transfers is hard
18:01.07leifmadsenwho 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)
18:01.15z4nD4Rhi .. mery x-mass
18:01.19leifmadsenwhen you transfer, there is no dialplan being executed, just channels being bridged
18:01.34leifmadsen(at least not dialplan that knows, "hey you're going to transfer a call!")
18:01.37z4nD4Rsomebody to help with voicemail?
18:01.47leifmadsen~ask
18:01.47infobotQuestions 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.
18:01.53paulc@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!)
18:02.11paulcZ4nD4R: What's up with voicemail?
18:02.11leifmadsenpresses option 4
18:02.29paulcOption 4 = Fetish Categories..
18:02.30leifmadsenpaulc: I'm building a psychic network right now... it's been a nightmare :)
18:02.31paulcyou sure you want to go there
18:02.39leifmadsenpresses *
18:02.55paulcTo talk to our hottest fantasy women, press 1...
18:03.05z4nD4Rpaulc: i have set voicemail ... http://www.asteriskguru.com/tutorials/asterisk_voicemail.html ... but dont send me mail about voicemail
18:03.09paulcPsychic eh? Similar kind of stuff to this project then.. (maybe we should be collaborating!)
18:03.15leifmadsenI 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)
18:03.39leifmadsenok, I gotta start this roast, so I'll be back in a bit :)
18:04.03paulcz4nD4R: 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)
18:04.18paulc@leifmadsen: Cheers - see you in a bit :)
18:05.29z4nD4Rpaulc: dnot know how to send email from commandline.... but mail should by set right
18:07.01z4nD4Rpaulc: 71001 => passw ,name , mail@gmail.com ; my record in voicemail.conf
18:09.41z4nD4Rpaulc: you see som error?
18:09.53paulcz4nD4R: Yeah, something like:   71001 => 1234,John Smith,john@acme.com
18:10.24paulcz4nD4R: Your line looks ok to me. Have you restarted Asterisk since changing voicemail.conf?
18:10.36z4nD4Roffcourse
18:10.55paulcz4ND4R: 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.
18:11.31z4nD4Ri started softphone.. and i becomme notification about voicemail
18:12.02z4nD4Rpaulc: how to make mail from command line?
18:12.42*** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
18:24.15Kattyhttp://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)
18:24.47Kattybut what should i put under the cheese?
18:24.51Kattythinking veggies.
18:25.48Qwellthe answer is obvious
18:25.50Qwellmore bacon!
18:25.59*** join/#asterisk m_tadeu (~quassel@89-180-98-120.net.novis.pt)
18:26.01Qwellactually, ground beef probably would go well with that
18:26.20theharBACON BACON BACON
18:26.22wizard171Eggs, Potatoes ...
18:26.41Qwellwizard171: that would work too.  breakfast...thing.
18:26.50Qwellbacorito
18:27.00_Corey_looks healthy to me
18:27.16Kattydies
18:27.19Kattyoh
18:27.19Kattyegg
18:27.21Kattyquiche, yes
18:27.23Kattyi like this idea
18:27.37Kattyupvotes Qwell, again
18:28.28paulcz4nD4R: delayed reply, sorry...  type "mail john@acme.com" (using your email address) from the command line and see what happens
18:29.04z4nD4Rpaulc: command line form asterisk or from linux console?
18:29.31paulcz4nD4R: 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
18:30.33z4nD4Rpaulc: i dont have "mail" instaled... i can say it is pure install ... only asterisk is installed
18:31.01leifmadsenor install mutt
18:31.04leifmadsenwhich I find easier to use
18:31.15Kattyalso likes mutt
18:31.31z4nD4Rleifmadsen: so i muss set in voicemail?
18:31.39z4nD4Rvoicemail.conf ?
18:32.10leifmadseneh?
18:32.18leifmadsenyou were talking about 'mail' command
18:32.54z4nD4Rleifmadsen: yes.. but it is deeper :D why my asterisk dont send mail about voicemail
18:33.25Kattyz4nD4R: have you confirmed it sends emails. period.
18:33.47z4nD4RKatty: ?
18:34.54Kattyi'll take that as a no.
18:35.13Kattyz4nD4R: you need to make sure your asterisk machine can properly send emails before working with the voicemail app.
18:35.34Kattyz4nD4R: as the voicemail app is dependant on the server's ability to properly send mail.
18:36.01z4nD4RKatty: yes... but how??? which program? mail? mutt? which is use by asterisk?
18:36.10leifmadsenz4nD4R: 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?
18:36.13leifmadsensendmail
18:36.16leifmadsen(or postfix)
18:36.34leifmadsenthen you have to check /var/log/maillog to verify it is sending email
18:37.09Kattyz4nD4R: i use mutt to do that.
18:37.16Kattyz4nD4R: to test emails.
18:37.22Kattyz4nD4R: mutt is simply an email client.
18:37.24*** join/#asterisk bmg505 (~leon@196-209-7-245.dynamic.isadsl.co.za)
18:37.41Kattyz4nD4R: you can send an email from command line using mutt, outside of asterisk.
18:38.10z4nD4RKatty: ok .. u have thi in voicemail? send mail with mutt?
18:38.19Kattyz4nD4R: mutt is an email client.
18:38.29Kattyz4nD4R: asterisk does not use mutt.
18:38.44z4nD4RKatty: what use asterisk to send emails?
18:38.48Kattyz4nD4R: i believe asterisk uses sendmail.
18:39.01Kattyz4nD4R: i use mutt to test from commandline
18:39.02leifmadsen<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?
18:39.29leifmadsenquestion asked and answered
18:39.46z4nD4Rok w8 .. i go try it mutt and sendmail.. and report :)
18:39.59Kattymutts leifmadsen
18:40.00leifmadsenchooses not to wait, and goes to eat lunch
18:40.08Kattyleifmadsen: enjoy.
18:40.55paulcI quite enjoy reading Kyle Rankin (or is it Shawn Powers?) go on about mutt in Linux Journal
18:41.08paulcI use it once in a while too..
18:41.15paulcit'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
18:44.00paulcYeah, 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
18:47.52paulcdear sox, telling me "pan is deprecated" is nice, but how about a hint of what to use instead? *grumbles*
18:50.53z4nD4RKatty: paulc 1. i can send email from linux console .. troght mutt
18:51.42*** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey)
18:52.11paulcz4nD4R: using the same email address as in your /etc/asterisk/voicemail.conf, right?
18:54.03z4nD4Rright
18:55.33paulcz4nD4R: Hmm.. how about voicemail contexts? you're leaving a message in the right mailbox/context that has the email address set up?
18:55.54z4nD4Rbut asterisk dont use mutt.... in voicemail.conf... isee mailcmd = sendmail -t ( bot comment thi line )
18:56.40Kattyfeels like naptime
18:57.53paulcz4nD4R: so have you tested sending mail using sendmail?
18:58.12z4nD4Rpaulc: no...only using mutt
18:58.27z4nD4Rif i understand.. i have 2 options
18:58.39z4nD4Rset voicemail.conf to use mutt
18:58.43z4nD4Ror install sendmail
18:58.46z4nD4Rright?
18:59.33paulcz4nD4R: no.. because mutt is a user program for sending mail, whereas sendmail is a mail transport.. MUA versus MTA or something
18:59.37paulcmutt was just to test if mail went through
18:59.44paulcit did, but we don't know if it used sendmail or went direct etc
19:00.01paulcfrom the commandline, can you do "mail john@acme.com" or "sendmail -t john@acme.com"
19:00.03z4nD4Rso i muss install sendmail..
19:00.18paulcif you don't have sendmail installed, then yes.
19:00.36z4nD4Rok
19:00.37QwellI would be shocked if you didn't have sendmail.
19:01.23makafreTo what does the Resource field correspond when we do jabber show buddies ?  Its right under the Buddy name...
19:01.38z4nD4Rpaulc: ok installed
19:01.51Qwellmakafre: it means nothing.  it's just a (user-configurable) identifier
19:02.05Qwellpeople use Home, Work, Tacos, etc
19:02.19z4nD4Rpaulc: i need some setting to sendmail?
19:02.27makafreQwell,  ok, I am asking because when I try to get JabberStatus it complains about the resource being null
19:02.39Qwellnull is probably not valid
19:02.58QwellI have no idea how to set/use it in Asterisk.
19:03.02paulcz4nD4R: I don't understand the question
19:03.10makafreok
19:05.23makafreabout the sendmail thing, I always use  sendEmail, it s a script and its easy to use, I prefer it over sendmail
19:06.00makafrehere is link to get it: http://caspian.dotconf.net/menu/Software/SendEmail/sendEmail-v1.56.tar.gz
19:06.23z4nD4Rpaulc: i have installed sendmail....  some additional options??in voicemail.conf.. or?
19:06.24QwellAsterisk doesn't need easy
19:06.33Qwellz4nD4R: No, sendmail requires no configuration.
19:06.58z4nD4RQwell: ok i try it sendmail -t my@email.com
19:07.06z4nD4Rbut nothing to do
19:09.07z4nD4RQwell: but in voicemail.conf i have commented line ;mailcmd = /usr/bin/sendmail -t .. uncommented this?
19:09.21QwellNo.  Do nothing.  Leave a voicemail, and it will just work.
19:09.40citywoks/will/should/
19:09.59QwellDo not question The Qwell!
19:10.11citywokI didn't question the qwell.  I just corrected him :P
19:10.17Qwelloh snap
19:10.35citywoklmao
19:10.50citywokgood thing russellb isn't here or i might have gotten KB'd for that
19:10.55z4nD4RQwell: but not work :D
19:11.19z4nD4RQwell: sry... it works :D
19:11.33Qwellcitywok: see? :P
19:11.44citywokhahaha, no i didnt :P
19:11.51citywoknothing to see here, please move along
19:12.09Qwellz4nD4R: so, fixed?
19:13.05z4nD4RQwell: yes.. i becomme mail with text and the voice messenge.. :) thx.. i muss only edit the text.. and it is ok..
19:16.55QwellI'm not sure what that means, but alright.
19:17.22z4nD4RQwell: text which is send in mail
19:17.39Qwelloh, okay.  thanks for clearing that up.
19:19.19*** join/#asterisk jetlag (jetlag@pool-173-61-245-217.cmdnnj.east.verizon.net)
19:27.28makafrehow to call a dialplan function from a perl AGI?
19:27.48Qwellsame way you would use a variable
19:27.56makafreok, 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
19:37.06makafreanyone knows who is the sender parameter for the JABBER_STATUS function?  (ref: http://www.asterisk.org/docs/asterisk/trunk/functions/jabber_status)
19:37.49makafrei know the resource and buddy
19:38.16makafrebut the sender?
19:38.33Qwellthat would be you
19:38.44makafreme?
19:38.56Qwellyour account
19:39.18QwellYou can have multiple accounts on a box, so you need to specify who you want to send from.
19:39.18makafreok, isn't it the same as the resource? I mixed up
19:40.19makafrelet's say I have in jabber.conf  [tacos] connecting as username=admin,   which one is the resource and which one is the sender?
19:44.54makafreok, I guess you mean they can be the same if there is only one user defined
19:46.43QwellI have no idea what resource is for, in that context.
19:47.45makafreok, sure
19:48.33*** join/#asterisk oDesk (~f@188.53.70.113)
19:49.01makafrewell, it doesn't work with tacos, <buddy>
19:49.04makafrelol
19:49.38oDeski want to Set(CALLERID(name)=D63xxxxxx)  cut the D letter into this line where it show sets the caller ID
19:50.33p3nguinresource:
19:50.34p3nguinClient or transport Asterisk users to connect to Jabber.
19:51.11*** join/#asterisk luckman212 (luckman212@pool-96-246-172-198.nwrknj.fios.verizon.net)
19:51.59p3nguinIt makes sense to me that the resource is NOT the username.
19:52.04makafreok, so If I have    [tacos] username=admin@xxxxxxx.com    "tacos" would be the resource
19:52.37p3nguinSince resource isn't the username, there aren't too many other choices.
19:52.49Kattyhi
19:53.00p3nguinhi2u
19:53.01QwellKatty: ohai
19:53.10KattyQwell: your turn for napping
19:53.29makafreI tried   with      admin@xxx.com, buddy@xxx.com, tacos    but it doesnt work, unless I have a bug in my AGI
19:54.24wizard171No, "username=admin@host/resource" where "resource" = something like "home" vs "work" etc.
19:54.59makafreah
19:55.10p3nguinI often use my host's name as the resource on my client.
19:55.31p3nguinI think Psi does that by default.
19:56.12wizard171ah, but do your ... resources ... need hosting?  (that is the question?) eh?
19:56.34p3nguinI have no idea what you're even trying to say.
19:56.41wizard171;)
19:56.58makafrelet me try that
19:57.56*** join/#asterisk ickmund (~ickmund@c-9149e755.015-144-70697410.cust.bredbandsbolaget.se)
19:58.01makafrequestion, are we talking about the username's resource or the buddy's resource?
19:58.27p3nguinA very good question that is.
19:58.34makafrelol
19:59.02p3nguinI'm not even sure what you're trying to do.
19:59.51makafreI 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
20:00.33makafreI was able to do it using JabberStatus but since it's deprecated I wanted to go with the function instead
20:02.10Kattysomeone lend me 75 cents
20:02.13Kattyi need caffeine
20:02.21leifmadsentosses Katty a quarter
20:02.49Kattyleifmadsen: i'm about to run to the store and buy a case of mt dew
20:02.56leifmadsenwell done
20:03.00theharexcellent
20:03.08Kattyhide them in the bottom drawer of the fridge
20:03.17Kattyno one else looks down there in the veggie drawer but me ^___________________________^
20:03.17theharnice
20:03.31theharnom
20:05.02*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:05.10leifmadsenmakafre: show us how you're using JABBER_STATUS()
20:05.24makafreok, give me a minute
20:05.43Kattywonders
20:05.49theharif
20:05.52Kattyif i can nap on the clock, i should be able to support local economy on the clock
20:05.55Kattyright??
20:06.00leifmadsenSet(JabberResult=${JABBER_STATUS(lmentinc,pbx@leifmadsen.com)})
20:06.06p3nguinWe has no one configured online payments for soda machines yet?  I could paypal your vending machine and you could go pick up your drink.
20:06.24Kattyp3nguin: that, sir, is amazing idea.
20:06.27leifmadsenmakafre: there is an example
20:06.37p3nguins/We/Why/
20:07.09citywokhmm. is chan_gtalk still allowing calls to PSTN?
20:07.18Kattyi bet i could paypal my company
20:07.23Kattyand they'd get me a soda ^_^
20:08.40Kattytada!
20:08.42Kattyafks
20:08.49leifmadsencitywok: ya as long as you update to 1.8.1.1 or later
20:09.01citywokhmm. okay i've got 1.8.1.1, then that didnt like me so i went to trunk.
20:09.07leifmadsenodd
20:09.13leifmadsenfollowing the docs?
20:09.44citywoksome random tutorial i google foundified
20:09.59makafrehere it is: http://pastebin.ca/2027523
20:11.16makafrebasically I call extension 10 which sends the "Incoming call" text to Spark, and then I am trying to send the status the same way
20:11.19citywokattempting the wiki.aster now, i was 95% of that
20:13.34leifmadsencitywok: make sure you use the one on the asterisk wiki
20:13.39leifmadsenah, you said that
20:13.57leifmadsenya the asterisk wiki is going to be the most up to date (Malcolm keeps it pretty up to date)
20:14.06Kattymmmm, caffeineee
20:16.27Kattyhas anyone planned christmas dinner yet?
20:16.39leifmadsenI plan to eat whatever my mom makes :)
20:16.54QwellKatty: whatever you make and then bring me :p
20:16.55WIMPyExcellent choice :-)
20:18.01KattyQwell: candied ham, scalloped potatos, broccoli
20:19.23*** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net)
20:21.01makafreleifmadsen: so you have  [lmentinc]  username=pbx@leifmadsen.com  in your jabber.conf
20:25.48leifmadsenmakafre: no, I have [lmentinc] defined and pbx@leifmadsen.com is the jabber user I'm checking the status of
20:26.22makafreok
20:27.48*** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au)
20:35.54makafregrrrr, problem solved, code 13
20:35.58makafrelol
20:36.15Kattyi'll code your 13 in a minute.
20:36.25makafrehaha
20:37.11WIMPyDo we hace to do metric conversions on error codes now?
20:38.00makafremy problem was in the way I was calling get_variable
20:38.32makafrenow is that moutain dew still available  ;-)
20:38.41Kattyit's in the veggie drawer.
20:41.15makafreget_moutaindew("veggie drawer");
20:43.31*** join/#asterisk riscphree (~riscphree@h235.10.91.75.dynamic.ip.windstream.net)
20:43.33Kattycarrar: ooooooooh
20:43.34Kattycarrar: btw
20:43.39Kattycarrar: got your card ^_^
20:44.02Kattycarrar: it's on my fridge.
20:45.24p3nguinis suddenly reminded to check the mail
20:47.54*** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au)
20:51.52CaneToadI 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
20:52.41CaneToadthe phone is displaying the correct caller id, but unlike when registering with the provider directly, somehow the extension number is displayed too
20:55.28WIMPyHas anyone here ever had any success with DAHDIs inbanddisconnect=yes? I only get random audio.
20:57.26WIMPyWith random=either of busy, silence or a dialtone.
20:57.42ChannelZDarn.  I was hoping you meant showtunes and rap
20:59.13Kattyhi ChannelZ
20:59.16Kattyhugs ChannelZ
20:59.25ChannelZHey Katty! LTNS
20:59.28ChannelZMerry Christmas
20:59.33Kattyty (=
20:59.38Kattywhat you been up to
20:59.57ChannelZNothing too exciting, just keeping busy with work
21:00.00ChannelZYou?
21:00.16Kattygoing through tons of wrapping paper
21:00.21Kattybut it's going to be so incredibly worth it
21:00.40ChannelZheh yeah that's one of my many jobs later tonight
21:00.46Kattyclose friend of mine has been going through a very difficult time lately. single mother, living at home with her father.
21:00.54Kattyworking to put herself through college
21:01.00ChannelZGotta finish cleaning the house, decorating, have a cheesecake to make... wrap...
21:01.12QwellI want a cheesecake.
21:01.17Kattythey don't have a lot of money, so i have decided to spoil the be-jeesus out of her son
21:01.25ChannelZ:) that's cool
21:01.43Kattyi think i'll take my camera over and make a video of it
21:01.53ChannelZQwell: unless you live in Colorado, you're out of luck :)
21:02.03Kattymm cheesecake.
21:02.05ChannelZWell.. not completely, but at least of mine.
21:02.13Kattymy cheesecake recipe is so easy...
21:02.16Kattyno baking required
21:02.24Kattyit's on the back of the gelatin box
21:02.33Kattydon't tell tho, everyone thinks its amazing ;)
21:02.39Qwelljello cheesecake?
21:02.45Kattyno, gelatin packets.
21:02.45ChannelZI've had one of those a long time ago I think
21:02.57KattyQwell: it's in the baking aisle.
21:03.14Qwelldoes cheesecake usually use gelatin?
21:03.23ChannelZNo
21:03.51Kattyusually you bake it in a spring form pan
21:03.52ChannelZSpeaking of food I need to go get lunch. BBL
21:04.08Kattybut the knox gelatin has a recipe on the back where you through 3 or 4 things in the blender with cream cheese and gelatin
21:04.12Kattythen chill it.
21:04.15KattyPOOF insta-cheesecake
21:04.26Kattys/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)
21:13.20GhOnDiEhi 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
21:13.29Kattyaww.
21:13.32Kattymy boss told me to go home
21:13.38GhOnDiEwhen looking at the sip debug it comes up with a 401 error
21:13.56GhOnDiEim guessing its a NAT problem but im not sure how to solve it
21:13.58z4nD4RKatty: happy man :D
21:14.42Kattyz4nD4R: he can be
21:14.50z4nD4RKatty: :D
21:14.56QwellKatty: Go home!
21:15.23Kattyi'm almost done
21:16.02Kattywhat is the asterisk abbreviation for thursday
21:16.40Kattythu?
21:16.57Qwellprobably
21:17.23Kattylet's hope so!
21:18.07Kattywell it didn't squeal at me.
21:18.27Kattyhope everyone has an awesome weekend if i don't see you before monday (=
21:18.31KattyQwell: especially you
21:18.41Qwell<3
21:18.50wizard171Katty:  Merry Christmas ... !
21:35.49*** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net)
21:37.48tymanAnyone 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
21:40.00tyman_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
21:40.31paulcI'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)
21:41.42tyman_Corey: Yeah, lumenvox looks cool.  Heard good things about it.  Just looking to do the gvoice type email/jabber of vmail
21:42.21paulcspeaking 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
21:42.49tymaneasy 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
21:43.00paulclike 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
21:44.00tymanhttp://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
21:46.03tymancouldn'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
21:46.46tyman_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***
21:47.12tymanyeah, more of a novelty in most cases...
21:47.46_Corey_but JUST accurate enough to be of utility
21:48.09tymangoog-411 is gone now...that didn't work to well for me for some reason...
21:48.15tymans/to/too/
21:48.58p3nguinI 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
21:49.45ChannelZWTF? You can only buy $50 max gift codes from iTunes?
21:50.35_Corey_ChannelZ: Can't beat Amazon mp3 store
21:51.07tyman_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. :-)
21:51.31theharperks up
21:51.32theharwhat?
21:52.04tymanChannelZ: Amazon mp3...only way to go.  256kbps vbr, with no horrible DRM
21:52.07ChannelZ_Corey_: yeah my personal choice but I was getting a couple gifts for clients who are in more or less apple fanboys
21:52.14theharKatty: 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
21:52.46ChannelZtyman: thankfully iTunes has mostly given up on DRM but regardless I try to not personally buy anything from them
21:53.36tymanChannelZ: 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)
22:05.55sthonis there any benefit to having hardware echo canceller?
22:06.28*** join/#asterisk infratel (infratel@154.5.144.132)
22:07.28ChannelZif you have lots of channels needing it at once, sure
22:09.03jShafinstalled asterisk (for the first time) on mac (via homebrew)
22:09.05jShaf./configure --prefix=/usr/local/Cellar/asterisk/1.6.1.6 --localstatedir=/usr/local/var --sysconfdir=/usr/local/etc --host=x86_64-darwin
22:09.11WIMPyThey might have a better algorithm and they save CPU time.
22:09.15jShafand yet i dont see any file in /usr/local/etc/asterisk/
22:09.15sthonChannelZ: 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.
22:09.18jShafshould there be one?
22:09.20infratelquick 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})
22:09.20infratelexten => _91NXXNXXXXXX,2,Dial(Zap/g2/${EXTEN:1})
22:09.20infratelexten => _91NXXNXXXXXX,3,Congestion
22:09.34infratel?
22:09.52tymanjShaf: did you make, make install?
22:09.59_Corey_sthon: I've found the quality of the HW echo cans better than software alone
22:10.03infratelg1 drops the out call to g2 out?
22:10.24sthon_Corey_: Thanks.
22:10.31_Corey_sthon: the tone thing is probably a fax
22:10.33jShaftyman: I see that the homebrew script does make && make install
22:10.38WIMPyinfratel: You should check DIALSTATUS and/or HANGUPCAUSE.
22:10.47_Corey_sthon: when a fax tone is detected, the echo can automatically should disable
22:11.15sthon_Corey_: that's probably the case, it doesn't happen that frequently.
22:11.18infratelcheck dialstatus? a ordinary user/client would not be required to do that
22:11.36tymanjShaf: if there were errors with make then make install may not run...Did you have build errors?
22:12.09WIMPyinfratel: ... in your dialplan. Befor trying again on the next interface.
22:13.13infratelWimpy, its actually not my dialplan. Just a web site does not explain the  grouping of two itentical dial plan lines with seperate group numbers.
22:13.47infratelIt 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)
22:18.11jShaftyman: I just now re-run the script, this time with verbose switch
22:18.16jShafI dont see any error
22:18.33jShafbut now I wonder if it is because I did not run "make sample" ?
22:18.38jShafthus no config files?
22:23.07WIMPyinfratel: 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)
22:30.42infratelokay that was the answer I was looking for.
22:31.16infratelCall it cascade channel aviability
22:31.25*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
22:32.14WIMPyThat's why I suggested chacking DIALSTATUS and/or CAUSECOSE.
22:32.55WIMPycaause=34 would seem obvious. Not sure it there are others.
22:33.24WIMPyMaybe 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)
22:41.59shmaltzwhy 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)
22:44.14ChannelZDid you build * yourself or is it from a package?
22:44.34shmaltzChannelZ, myself
22:45.16ChannelZhmnm.  All of it at the same time?
22:45.28shmaltzChannelZ, all of what?
22:46.01ChannelZAsterisk.  Like you didn't compile it and then a month later compile a part of it for a different module or something
22:46.52shmaltznope, all at the same time
22:47.01shmaltzmake, make install, make configs
22:47.19ChannelZhmm
22:48.40shmaltzshould I recomplie just that folder?
22:49.41wizard171shmaltz: sounds like you are missing "autoconf" as a package during build, or you do not have "libcap.so"? (which, would be very strange)
22:50.10ChannelZyeah or has something changed?  (did it ever work?)
22:50.21shmaltzChannelZ, I doubt it ever worked on this machine
22:50.40shmaltzwizard171, is it possible to compile anything without autoconf?
22:50.46shmaltzchecking now for libcap.so
22:51.02shmaltzis libcap.so part of linux or asterisk?
22:51.23ChannelZYeah I'd figure if you were missing something it'd have just failed.  Maybe your libcap is a crazy old version or... I dunno
22:51.31shmaltzroot@pbx:/# ls -R | grep libcap.so returns some results
22:51.34ChannelZit's separate
22:51.59shmaltzlibcap.so verison is 2.19
22:53.12ChannelZhmmm
22:54.26wizard171shmaltz: 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 ...
22:54.29shmaltzit appears taht I'm getting on a few modlues
22:54.44shmaltzI'm going to reinstall libcap.so
22:54.51ChannelZdo you have libcap-dev or whatever your distro might call dev packages
22:56.00wizard171ah, "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 ...
22:56.18shmaltzI'm running 1.4
22:56.24wizard171s/makefiles/otherstuff/
22:57.05shmaltzwizard171, how do I "link" to it?
22:57.50wizard171ordinarily, 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. ...
22:58.00ChannelZheh interesting, looking through one of my config.log from a previous build, the lcap test failed
22:59.28shmaltzhow do I know if I have bison/flex?
22:59.55shmaltzconfig.log reports both bison and flex
23:00.43NightMonkeyGosh I heart asterisk.
23:01.07NightMonkeyI just moved my home phone system off a troublesome power-hungry server to a Sheevaplug. :)
23:02.11shmaltzan identical config (other than kernel) is running music on hold fine
23:03.25wizard171libcap is part of "posix" ... and res_musiconhold ... forks ... perhaps your "posix" is one of the ... other implementations?
23:03.29shmaltzwhat is the name of the tool that relinks lib?
23:04.02wizard171"ld"
23:05.04shmaltzdidnt help :'(
23:05.11shmaltz:o
23:05.41wizard171shmaltz: 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 ?
23:05.55shmaltzslackware 13.1
23:06.15wizard171and you are using "yum" or ???
23:06.27shmaltzpkgtool
23:07.50shmaltzis recompiling from scratch
23:10.01shmaltzrecompiling didnt help :'(
23:11.40wizard171grab the latest "autoconf" from "ftp.gnu.org" and install it, then try it again ...
23:11.43shmaltzcould it be a kernel options that I'm missing?
23:12.32*** join/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp)
23:12.49dongshi, im using asterisknow i need to add a new IP to the machine wiht it, how do i do that.
23:13.27p3nguinThis is the Asterisk channel, not the AsteriskNOW channel.
23:14.15p3nguinIn here, we add IP addresses to interfaces with 'ip' or 'ifconfig'.
23:14.21dongsyes, it also has 195 people (not counting you as a person) and the other one has 11 bots.
23:14.35dongsso asking here is much more likely to get a response
23:15.01dongsp3nguin: 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)
23:15.34p3nguinIt's CentOS, and all the pertinent information is found in the AsteriskNOW quickstart guide.
23:16.51p3nguinI'd be somewhat surprised if you didn't have to set the root password during setup.
23:17.17shmaltzwizard171, 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)
23:19.44shmaltzdongs, just boot to userlevel 1 and change it
23:19.55shmaltzor dual boot and chroot and run passwd
23:20.26dongsyeah, i did remember setting password and my default one worked.
23:20.33dongsinterface added, problem solved
23:20.39dongsnow i can go ask freepbx guys all stupid questions.
23:20.50shmaltzdongs, :-h
23:21.34dongsthanks
23:21.35*** part/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp)
23:28.04wizard171shmaltz: strange ... in the source directory, for asterisk, you may need to "make clean" ... then "configure/make" again ...
23:28.39shmaltzwizard171, I deleted the whole directory before recomplining
23:28.43shmaltzdoes that count?
23:29.13wizard171:) ... yep! (but, my brain is working on the possibles??)
23:29.30shmaltzok, so I should still do make clean?
23:30.11wizard171Not if "autoconf" was there ... before ... "configure/make" ...
23:30.27wizard171on a fresh directory ...
23:31.03shmaltzwizard171, restarting the machine, lets see if that helps (i doubt it, but trying)
23:31.30shmaltzis rooting a nookcolor at the same time
23:31.59wizard171okie, dokie, then ... sometimes I hear that works on ... windows ... as well ... ;)
23:32.33shmaltzok restart gave me autconf 2.68
23:32.40shmaltzlol
23:33.33shmaltznow recomplining
23:33.55wizard171*that* should be ... interesting ...
23:34.30*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
23:36.20shmaltzstill no good :'(
23:39.03shmaltzwizard you think asterisk 1.4.38 will be better than 1.4.37?
23:39.41wizard171ah, some reason you wanted 1.4 ... vs any other ?
23:40.13shmaltzwizard171, like 1.2?
23:40.35wizard171heaven forbid! ... how about 1.6.2.13?
23:41.04shmaltzwizard171, dont push it, I have been using 1.2 till like 3-4 weeks ago, why jump so high?
23:41.37shmaltzmost of my installs are STILL running 1.2
23:41.46shmaltzi might still have one 1.0x somewhere
23:42.16wizard171you 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)
23:42.44shmaltz:)
23:46.04shmaltzof course 1.4.38 didn't help
23:46.17shmaltzok, posting to lists
23:46.45wizard171ah, how bout a "pastebin" of the "logs" from "configure/make" ...?
23:47.00shmaltzok, my sis in-law just emailed me this:
23:47.02shmaltzDear 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
23:47.14shmaltzwizard171, is that config.log?
23:47.36wizard171yes ...
23:49.14paulcshmaltz: your sister makes me laugh - I can relate :)
23:49.47WIMPySister in alaw or sister in ulaw?
23:49.53shmaltz:D
23:49.56paulcChanSpy - 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?
23:50.04paulcWIMPy: *groan* that's awful!
23:51.41wizard171paulc: yes, unfortunately, until 1.8 ... then you can give arg to have it do just one and exit ...
23:56.30wizard171shmaltz: I am not abandoning you ... I just need to go AFK for a few ...
23:56.44shmaltzthanks
23:56.49*** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com)
23:57.05shmaltzwizard171, here is the pb: http://pastebin.ca/2027732

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