IRC log for #asterisk on 20111224

00:19.33*** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net)
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00:37.01voipengso asterisk -rx 'core show channels' | tail -f should tail the command?
00:37.38WIMPyNo. The output.
00:38.02lanningnix the "-f"
00:38.40WIMPydoesn't think it would do anything.
00:39.21voipengheh that didnt work eith
00:39.23voipengeither*
00:39.44voipengwhat do you mean the output?
00:40.47WIMPyYou can't tail the command.
00:40.58voipenggotcha...
00:41.00voipengthat sucks
00:41.25voipeng<PROTECTED>
00:41.36WIMPyhuh?
00:41.46voipengrun the command automatically with a crontab
00:42.23WIMPyWhat are you on about?
00:42.34voipeng??
00:44.09voipengps, you can just get a notepad, enter the command, enter an extra line, and copy multiple times, it will just continue to make the command run
00:44.37voipengjust figured that out hah
00:53.08p3nguinhuh?
00:53.30p3nguinYou want to see the command run time after time after time?
00:53.43voipengyea works for me
00:53.52kaldemarwatch is the key
00:54.28p3nguinwatch -t 'asterisk -rx "core show channels"'
00:54.53p3nguinNeed to see it more often?
00:54.54p3nguinwatch -n1 -t 'asterisk -rx "core show channels"'
00:55.39p3nguinWant to see changes get highlighted?
00:55.41p3nguinwatch -n1 -d -t 'asterisk -rx "core show channels"'
00:55.47voipengniceeee :)
00:56.36p3nguinI don't even understand the notepad thing you were trying to pull off.
00:57.34voipengthat last command is awesome
00:57.42voipengheh i just entered the command a ton of times
00:57.50voipengand let it keep running lol
00:58.57voipengthe highlight is cool but if you have a bunch of changes it sorta is useless
00:59.36p3nguinEverything will get highlighted, probably.
00:59.50voipengyea and it keeps extending in the window
00:59.55voipengbreaking the information apart
01:00.28p3nguin`watch` has some other options, too.  Check the man page if you're interested.
01:00.34voipengwill do thanks
01:04.36voipengadding cat -v to the end of the command works tho
01:04.50SeRip3nguin: new phone is here.
01:04.58SeRiupdating app now
01:05.09SeRiis very nice and clean
01:05.49SeRihaving a back lit screens makes the world a lot nicer
01:08.59*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
01:10.47voipengdamn clients shouldnt be working tonight, want to do my update and leave
01:11.30p3nguinAny awk gurus around?  I need to grab just the UUID out of this line:  "OpenBSD" {6da3edfd-2b61-49dd-c2ab-2b65b355b983}
01:12.21p3nguinBasically, I just need to grab the string between { and }.
01:13.18lanningsed -e 's/^[^{]*{//' -e 's/}.*$//'
01:14.24p3nguinThat will certainly be an option if I can't do it with awk.
01:17.10*** join/#asterisk kaushal (~kaushal@115.118.229.80)
01:18.19kaushalHi
01:18.44kaushalwhat all technology is being used in Asterisk
01:19.03kaushalis it sip or iax2 or dahdi or zap and any other ....
01:19.14kaushalAny documents or wiki page
01:19.50kaushalwhat are the differences between them and which technology to use in what context
01:21.28kaushalAlso is there a flow chart when any user dialin to an inbound number or dials out using an outbound number ?
01:21.33WIMPyAll of those and even more.
01:21.57kaushalI mean what happens behind the scene
01:22.08WIMPyYour dialplan.
01:22.40kaushalWIMPy: any flow charts ?
01:24.15WIMPyYou make them.
01:24.27voipengorigination provider -> Sip trunk -> NOC -> SBC -> FS -> CPE
01:25.02kaushalSBC and FS ?
01:25.51voipengsession border controller
01:25.53voipengand feature server
01:26.29kaushalvoipeng: Any wiki on asterisk ?
01:26.38voipengvoip-info.org :)
01:26.42kaushalok
01:26.46WIMPywiki.asterisk.org
01:26.47voipengi like digiums forums too
01:27.08kaushalvoipeng: can you please point me to the exact url or wiki or voip-info.org ?
01:27.18voipenghttp://forums.digium.com/
01:27.30voipenghttp://voip-info.org/
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02:02.56[TK]D-Fender~wikis
02:02.57infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
02:03.40[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Home
02:04.00[TK]D-Fender^  first place to look outside of the docs included in the tarball
02:04.59[TK]D-Fenderkaushal: SIP is a VoIP protocola, as is IAX2, both are quite different.  ZAP = DAHDI (name changed) and is a hardware interface layer for cards like Digium's, etc
02:05.05[TK]D-Fenderkaushal: This is all explained in ....
02:05.06[TK]D-Fender~book
02:05.06infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
02:05.09[TK]D-Fender^^
02:05.14kaushal[TK]D-Fender: ok
02:05.23[TK]D-Fenderkaushal: Which is what you should have been reading all these weeks you've been in here
02:05.59kaushal~buybook
02:05.59infobotYou can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY
02:08.03kaushal[TK]D-Fender: Thanks again
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02:43.31voipengso if i need to stop asterisk after a reboot and start it for it to come up... was that just the modu chan_sip reload?
02:43.46voipengis there a way to check the log to see why it didnt start on its own
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03:13.59p3nguinif i need to stop asterisk after a reboot and start it for it to come up   <---- does not make sense.  If you must stop it, it is already up.
03:14.34p3nguinIf you want to reload sip.conf, the command is "sip reload".
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03:43.31carrarvoipeng:
03:43.33carrar~book
03:43.33infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
03:44.20p3nguinha
03:44.32carrarWhat is that, 1st chapter?
03:46.39p3nguinWhere ever it is, it's too much trouble to learn anything about it, I'd guess.
03:49.33ChannelZThe first chapter of the last book you'll never read
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04:22.22EugeneKayWhat's a decent Windows SIP client that will integrate with Google Contacts(NOT Gtalk/XMPP), if such a beast exists?
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06:08.46SeRip3nguin: you around?
06:09.01p3nguinYes.
06:18.48SeRiI for got how to repace a database key
06:20.17SeRiI have a new phone with a new mac and want to repalce it and all I want to do is add the new mac to the database replacing the old one
06:20.20SeRip3nguin: ^^
06:21.03SeRidatabase put phones device 1003 machere?
06:21.10p3nguindatabase put phones YOUR-EXTEN/device SIP/NEW-MAC-ADDRESS
06:21.21SeRiah!
06:22.12p3nguinAs soon as you hit enter, your extension now points to your new phone.
06:22.31SeRinice!
06:22.32SeRi:D
06:23.22SeRihow you been?
06:23.41p3nguinHmm, I wonder if it should be:  database put phones/YOUR-EXTEN device SIP/NEW-MAC-ADDRESS
06:23.58p3nguinFair, I suppose.
06:24.49SeRiyou had it right
06:24.54SeRiI have family over from PR.
06:24.57SeRimy brotehr is in
06:25.42p3nguindatabase put phones/YOUR-EXTEN device SIP/NEW-MAC-ADDRESS  will put the value in the same place.
06:26.00p3nguinI was going for correctness rather than result.
06:26.15SeRicool. I see
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07:49.47SeRip3nguin: .....
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08:22.52SeRip3nguin: the phone uses g722 and so far so good :)
08:22.59SeRistill working some of the old config in...
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11:19.49BlackBishopany way I can make asterisk act as a client too ?
11:20.13WIMPyClient for what?
11:20.17BlackBishopI'd like it to register for incoming/outgoing calls @ another sip server which I have credentials and manage the in/out from my own asterisk
11:20.32WIMPyThere is no difference.
11:20.42WIMPyAnd to register see "register =>"
11:21.30BlackBishopthe thing is that I don't want to have 3 sip accounts set up on my clients ( phone/computer ) .. just 1 ( my asterisk ) .. and I'll make my own dialplans ..
11:21.42BlackBishopthanks, I'll look into register
11:21.45BlackBishopgoes google
11:23.33WIMPyThere's also "callbackextension" for an alternative way to register.
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15:47.31mawelshi
15:50.22mawelsis there any way to get celliax working with asterisk 1.6.2?
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15:58.50mawelsi want to connect my old mobile phone to my asterisk 1.6 .2 box but it hasn't bluetooth
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17:41.41[sr]helou my friends
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18:46.38Godfather_hi
18:47.44Godfather_i've installed 1.8.8.0 and when i connect to the cli i have no modules like sip, if i try to do "sip show peers" for example i get No such command 'sip show peers' (type 'core show help sip show' for other possible commands)
18:49.03Godfather_and if i do "module show like sip" i get no modules
18:50.13Godfather_i downloaded the tar.gz, did ./configure, make, make install, make config, and i run it
18:59.52cmendes0101asterisk -c, do you see errors outputing?
19:00.19[TK]D-FenderGodfather_: If that's all you did then you have no configs at all
19:00.29[TK]D-Fenderthat means no sip.conf, no modules.conf, etc.
19:00.35[TK]D-FenderNo modules could even load
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19:03.52bbourdageI would recommend a make samples
19:03.52psharmorCan anybody help me with this error:  JABBER ERROR: No DNS user_gmail_com for client to talk.google.com?  Google has provided nothing useful
19:05.42Godfather_[TK]D-Fender what i forgot?
19:06.06[TK]D-FendergodYour configs?
19:06.13Godfather_i dont want to run make samples
19:06.33[TK]D-FenderGodfather_: Well did you verify all the pertinent configs?
19:06.48[TK]D-Fenderasterisk.conf , sip.conf, codecs.conf , modules.conf, etc?
19:07.19Godfather_[TK]D-Fender what should i verify in a fresh install?
19:07.50[TK]D-FendergodDo you even have them?  Without 'make samples" on a fresh install you have absolutely nothing
19:08.12[TK]D-FenderGodfather_: You should be doing that at a bare minimum and then clear out the important configs.
19:08.23[TK]D-FenderGodfather_: because you are missing all the crucial basics.
19:08.42[TK]D-FenderGodfather_: no indications.conf also means no in-call progress, etc.
19:10.35Godfather_[TK]D-Fender ok, with "make samples" and restart asterisk i have those modules
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19:10.49[TK]D-FenderGodfather_: then go configure your system
19:11.01Godfather_[TK]D-Fender now its ok
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19:17.08Godfather_[TK]D-Fender the main problem was i have installed asterisk on a VPS, i tried 1.6.2.x but i get no audio (on calls, on musiconhold, etc). Now i tried with 1.8 and continues with no audio
19:17.30Godfather_[TK]D-Fender heres the cli output http://pastebin.com/rNdt2ibx
19:17.48[TK]D-FenderGodfather_: Did you think that the entire 1.6.2 branch was simply broken and that changing to 1.8 was a fix?
19:17.59Godfather_[TK]D-Fender no
19:18.17bbourdageGodfather, are you phones and the server connected to the same network ?
19:18.33Godfather_bbourdage no, the server and outside my network
19:18.40Godfather_*is outside
19:18.54Godfather_i set externip on sip.conf
19:19.15bbourdagehow many phones do you have locally ?
19:19.16Godfather_add the user that is calling to 91 has nat=yes
19:19.39Godfather_bbourdage i just trying exten => 91,1,MusicOnHold() to try the audio
19:19.48[TK]D-FenderGodfather_: Answer() the call first
19:20.11[TK]D-FenderGodfather_: And go prove that your musiconhold even has a working mode and files/etc to back it up
19:21.05Godfather_[TK]D-Fender the answer() before was the problem :-/
19:21.17Godfather_ty
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21:04.19carrarmoo
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21:24.15[TK]D-Fender~areyouadog ?
21:24.15infobotBark! Bark!
21:24.19[TK]D-Fenderinfobot: Good boy!
21:24.19infobotaw, gee, [TK]D-Fender
21:24.25[TK]D-Fender~botsnack
21:24.25infobot[TK]D-Fender: aw, gee
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21:59.08yamahar1MERRY CHRISTMAS! Haven't been on IRC in sometime.
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23:40.17p3nguinMary Crispness
23:42.55dijibMarry Christmas to you too p3nguin

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