IRC log for #asterisk on 20160425

00:00.05*** join/#asterisk fstd (~fstd@unaffiliated/fisted)
00:36.20*** join/#asterisk JavierLazo (446c18ce@gateway/web/freenode/ip.68.108.24.206)
00:36.29JavierLazohola alguien aqui que hable espanol???
00:36.49JavierLazopregunta si hay alguien que hable espanol y lo pueda ayudar
00:37.43hans-christianQuiero que esto no comprar la alfombra :>
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00:38.31JavierLazoHello, I need help from someone who can help me set my asterisk, and I gave up, I have four days on trying and I can not move
00:39.35hans-christianI add also some problem. And got directly some help :>
00:39.39hans-christianerm "had"
00:39.52hans-christianask you questions and do ask to ask
00:40.00JavierLazosomeone can help me with my settings sip.conf and my extensions.conf
00:40.22JavierLazoI have already registered the trunking
00:40.36JavierLazobut when I try to dial calls will not let me
00:40.44JavierLazoexample
00:40.45JavierLazo[Apr 24 16:34:52] NOTICE[9874][C-00000000]: chan_sip.c:25399 handle_request_invite: Call from '' (64.118.91.100:5060) to extension '5348450989' rejected because extension not found in context 'incoming'.
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00:52.31JavierLazothanks
00:52.49JavierLazorobink_:
00:52.54JavierLazohello
00:53.23JavierLazoi needdddddd heelllllppppppp
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00:55.59JavierLazohello
00:57.09*** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:e104:4854:dabc:cbf4)
00:57.56JavierLazoraspberrypifan:
00:58.02JavierLazohi
00:58.07raspberrypifanhi
00:59.42JavierLazoraspberrypifan:
00:59.54JavierLazoI need help
01:00.26JavierLazoI spent 4 days trying to configure my asterisk, and have failed
01:00.36raspberrypifanwhat happeend
01:00.45JavierLazoyou can help me please
01:00.47JavierLazook
01:01.26JavierLazoand record the trunk but now I need that incoming calls to my softswitch, who come from another server out my asterisk
01:01.27raspberrypifanwhat
01:01.48JavierLazoYou understand
01:01.50JavierLazook
01:01.55JavierLazoi have mi asterisk ready
01:02.02JavierLazothe sip trunk
01:02.04raspberrypifansi
01:02.10raspberrypifancuenta el problema de una vez
01:02.15JavierLazothe sip trunk
01:02.20JavierLazohablas espanol
01:02.21JavierLazook
01:02.35JavierLazotengo mi asterisk registrado con la troncal sin problema
01:02.37JavierLazolocalhost*CLI> sip show registry Host                                    dnsmgr Username       Refresh State                Reg.Time trunking.voipdnsservers.com:5060        N      665457388           31 Registered           Sun, 24 Apr 2016 17:05:25 1 SIP registrations. localhost*CLI>
01:03.06JavierLazoahora lo que necesito es que desde mi softswitch yo pongo como end point carriers mi asterisk o sea el ip puerto 5060
01:03.14JavierLazolas llamadas cuando entran a mi asterisk me da esto
01:03.42JavierLazo[Apr 24 17:07:16] NOTICE[10139][C-00000000]: chan_sip.c:25399 handle_request_invite: Call from '' (64.118.91.100:5060) to extension '5348750989' rejected because extension not found in context 'incoming'.
01:03.49JavierLazoy no conecta la llamada al destino
01:04.10JavierLazoraspberrypifan: tienes idea como ayudarme
01:04.12JavierLazo?
01:04.33JavierLazoes problemas en mis config que no son buenas
01:04.37JavierLazoy necesito ayuda con eso
01:04.43raspberrypifanpero copia la configuracion sample
01:04.45raspberrypifany ahi le cambias
01:04.49raspberrypifanel problemas es el contexto
01:05.10JavierLazollevo 4 dias y no logro que funcione
01:06.19raspberrypifancoppia el sip ejemplo
01:06.20raspberrypifany ahi busca
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03:51.34raspberrypifan[TK]D-Fender: hello friend
03:58.12raspberrypifani cant get yate to register with my asterisk vm install
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07:54.22tparcinaHow to specify default time zone in voicemail.conf?
07:54.43tparcinaSo that I don't have to specify it for every account.
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10:52.01davlefoutparcina, Bonjour, il faut configure des groupes selon les zones choisis.
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12:08.33amonktparcina: [zonemessages]
12:08.33amonkeastern=America/New_York|'vm-received' Q 'digits/at' IMp
12:08.33amonkcentral=America/Chicago|'vm-received' Q 'digits/at' IMp
12:08.33amonkcentral24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
12:08.33amonkmilitary=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
12:08.36amonkeuropean=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
12:09.02amonksee [zonemessages] in /usr/local/share/examples/asterisk/default/voicemail.conf or whatev on your system
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12:11.11tparcinaamonk: TY
12:11.22amonksure
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13:31.46Rasputin3711Hello
13:32.05Rasputin3711What is the best version Asterisk for production system ~50 users?
13:32.42[TK]D-FenderAny that is current and supported
13:32.48[TK]D-FenderSee !topic
13:32.55Rasputin37111.8.xx or 13?
13:32.59[TK]D-Fender13
13:33.01[TK]D-Fender1.8 is DEAD
13:33.17[TK]D-FenderTopic for #asterisk is: #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.2 (2016/04/20), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
13:33.48Rasputin3711I know, but i see a lot of topic then people don't recomende 13.8.
13:34.41[TK]D-FenderI'd be interested to see who, where, and why...
13:35.00Chainsaw11 is the lowest you can go, but for a new install, I'd go with 13.
13:35.04[TK]D-FenderThe only supported branches are 11 & 13.
13:35.11ChainsawIt's where the development effort is going.
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13:35.15[TK]D-FenderAnd 11 Won't be supported that much longer
13:35.45Rasputin3711Do you have systems on 13 ver with more than 100 users?
13:35.49fileyeah, I'd be interesting in seeing the reasons why people aren't using 13
13:35.50[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
13:35.54[TK]D-Fender100 users is nothing
13:36.24[TK]D-Fenderthey've pushed upwards of 10000 calls through a single server ( under controlled circumstances)
13:36.42Rasputin3711What OS?
13:36.45[TK]D-FenderLinux
13:36.50Rasputin3711Distro?
13:36.58[TK]D-FenderDoesn't matter
13:37.03[TK]D-FenderNo magical difference
13:37.39[TK]D-FenderAsterisk doesn't care.  It is either built properly and works, or was not and did not.
13:37.54ChainsawAnd an element of personal preference always creeps into distro choices.
13:37.57Rasputin3711Do i need update my system 13.7 -> 13.8.2 or it's not necessary?
13:38.10[TK]D-Fender13.8.2 came out for a reason
13:38.16ChainsawRasputin3711: Always go for the most recent version. Bugs get fixed all the time.
13:38.33[TK]D-FenderThere are obviously some mix of bug & security fixes, and possibly new features
13:38.48GuggeAnd new bugs :)
13:39.03Rasputin3711Where can i find bugs for certain version for example(13.8.2)?
13:39.05[TK]D-FenderI want only the newest, most fashionable bugs
13:39.09filewe've actually been pretty good
13:39.11[TK]D-Fenderin the changelog
13:39.22[TK]D-FenderIt's included in the tarball and on the websitte
13:39.33Rasputin3711After release
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13:40.16Chainsawfile: My downstream patchset isn't growing, which is a really good sign.
13:40.57fileour CI infrastructure is finding stuff before merging, which is really
13:41.21[TK]D-FenderThere is a bug tracker as linked off the main site
13:41.34filegood
13:41.36filemissed a word
13:41.38[TK]D-FenderFor current issues being examined
13:42.21[TK]D-FenderTypically new bugs aren't all completely new, they are just older issues being discovered.
13:42.52[TK]D-FenderSo going older means you should likely just have more: both those already fixed, and those that are common and in progress of being fixed
13:43.21[TK]D-FenderSo unless there is a very specific issue that doesn't hit a slightly older release then there is no sense in starting backwards
13:43.26Rasputin371113.6.0 has good bug with sip and i don't want to get it another time.
13:43.36[TK]D-FenderBecause that ignores things that have actually been fixed
13:43.54[TK]D-Fenderand was that bug fixed?
13:44.08[TK]D-FenderDo you see it re-issued?
13:44.28Rasputin3711It was closed in the next version 13.7.
13:44.38Rasputin3711But it was not fun in production.
13:45.24[TK]D-Fenderwas that bug only in 13.6.0?  Did it list as applying to earlier versions?
13:47.20Rasputin3711Yes, https://issues.asterisk.org/jira/browse/ASTERISK-25476
13:48.24[TK]D-FenderThink I remember that one coming up a few times
13:48.38[TK]D-FenderWas a bigger one....
13:48.57[TK]D-FenderSo keep an eye on the tracker and jsut start witht he latests
13:49.19Rasputin3711What is your rule how to update to new version?
13:50.39[TK]D-FenderTest it if you can, give it a few days to see if anything interesting hits the tracker, then do itt
13:50.46[TK]D-FenderBasically : common sense.
13:50.58[TK]D-FenderDon't be gun-shy about it
13:51.13[TK]D-FenderSeeing the changelog to see what's fixed is always a good idea as well
13:52.36filefrom a project perspective we test what we can, with older/less used stuff having less tests and the tests not covering wide range of usage
13:56.16Rasputin3711Many thanks!
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17:15.11drale2kHey all. I am new to Asterisk. I have set it up on my test ubuntu server and it works fine for internal use (iphone to mac inside the network)
17:15.43drale2kCan i now have the iPhone app configured externally (using my public server IP) and use it or is the ping / delay too high ?
17:15.54drale2kor is asterisk fine with 30-50 ms ?
17:18.33newtonrit'll probably work. it will however be highly variable in call quality
17:19.53drale2knewtonr: hm what is the better solution? I would like something that does not go over landline
17:20.36drmessanoI use an iOS softphone without too many issues
17:20.50drmessanoYou hit a bad spot and obviously youre stuck
17:21.08newtonrYeah I mean if you have a strong data connection it can work just fine
17:21.13drale2kdrmessano: is it just configured with the Asterisk Server IP ?
17:21.22drmessanoIt wont be _as_ reliable as your native calls on your phone
17:21.25drmessanoYep
17:21.35drale2kuseable is good enough for me
17:21.45drale2klike hear clearly
17:21.52drmessanoYes
17:22.05drmessanoIts actually been a lifesaver
17:22.20newtonrbe sure you have a strong password and user name if you are exposing on a public IP :) or filter by your phones MAC or something.
17:22.31drale2knewtonr: yup sure
17:22.35drmessanoI have a few locations where I have no cell service, but I do have Wifi
17:22.42drmessanoSo that works
17:22.50drmessanoand 100% what newtonr said
17:22.52drale2kdrmessano: ye, does it work well with 3G ?
17:23.03drmessanoWhen you say "3G"
17:23.11drmessanoDo you mean "3G" or are you being generic?
17:23.17drale2k3G the phone network
17:23.24drale2klike no WiFi but 3G
17:23.25drmessanoThats not what I am asking
17:23.48drmessanoMost places are 4G/LTE
17:24.04drmessanoAre you specifically talking about 3rd generation networks?
17:24.16drale2ki am asking 3G, not 4. the thing before 4G / LTE
17:24.23drmessanoAh
17:24.29drale2kbecause i expect it to work with LTE
17:24.49drmessano3G works.. I would say maybe 20% less reliable than 4G
17:24.56drmessanoThe 3G networks sucked
17:25.01drale2kby reliable you mean quality or break ups
17:25.09drale2klike disconnects etc
17:26.12drmessanoA Call dropping is not a quality issue, its a capacity issue
17:26.21drmessanoand thats obviosuly not an issue
17:26.29drmessanoand thats obviosuly not an issue with your own server
17:26.47drmessanoBut on 3G you may have jitter and could lose connection
17:27.00drale2klose connection is a call drop for me :P
17:27.04drale2kbut i get what you mean
17:27.10drmessanoNot necessarily
17:27.22drale2kcould it just reconnect ?
17:31.32*** join/#asterisk ElliotJohn (be352905@gateway/web/freenode/ip.190.53.41.5)
17:32.39ElliotJohnHi guys, I´m kind of new in this Asterisk business...need some help in choosing a box
17:34.36Synthase_Choosing a box?
17:36.24drale2ka phone probably
17:39.23ElliotJohn480 concurrent calls
17:39.53ElliotJohn.    600 user sites
17:40.21ElliotJohnsupport is required for FXO/FXS/PRI/BRI circuits
17:40.44ElliotJohnConcurrent calls to be supported + allowance for any CODEC transcoding to be done +  allowance voice recording to be done
17:41.11ElliotJohnDISC SIZE   =    500GB   +    voicemail space   +   voice recording space
17:42.15Synthase_Transcoding varies by codec, I wouldn't leave that open ended as some are more CPU intensive than others.
17:42.38ElliotJohnCDR/reporting can be handled, VM
17:46.30ElliotJohn2 x 10/100 NICs running initially in primary/secondary HA mode
17:49.51ElliotJohnso the cluster can always be grown at any future date if the requirements increase due to concurrent call  numbers >480 and/or heavy voice recording and/or heavy voicemail and/or complex transcoding
17:50.41ElliotJohnwhat brand/model do you guys recomend ??
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18:57.51drale2ki haved forwarded all ports properly but when i try to connect with a softphone from the outside, i get a "Wrong Password" error in the asterisk CLI
18:58.00drale2kit works fine internally
18:58.14drale2kits just for testing so i am using my public home IP
18:58.27drale2kasterisk server runs on my laptop
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19:13.02wyoungBacon pancakes! Making bacon pancakes! Get some bacon and I put it in a pancake.  Bacon pancakes, that's what I'm gunna make! Bacon pancakes!!!!
19:16.49*** join/#asterisk azerus (~badass@unaffiliated/badass)
19:16.52Nuggetwyoung: https://www.youtube.com/watch?v=cUYSGojUuAU
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20:43.12*** join/#asterisk CryptoManiac (~Crypto@41.222.28.166)
20:44.55CryptoManiacWhere's good old [TK]D-Fender ?
20:45.15CryptoManiacHe's always here, weird not to see him here
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20:46.07*** join/#asterisk infernix (nix@unaffiliated/infernix)
20:54.57wyoungNugget: :D
21:08.56*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
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21:48.26[TK]D-FenderCryptoManiac, Was in-transit home
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22:05.09*** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-05-1176475207.dsl.bell.ca)
22:15.06CryptoManiac[TK]D-Fender: Thought so. (That you will be away for a minimal time).
22:15.55CryptoManiacSomeone else needed some help with a jitter buffering issue so I referred him to you
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23:09.35*** mode/#asterisk [+o cresl1n] by ChanServ
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23:13.10*** mode/#asterisk [+o cresl1n] by ChanServ
23:16.32wyoungso I get a subscriber absent error when dialling out.  I restart asterisk and all is good :\
23:16.51wyoungAny way to tell asterisk to do a restart if this happens in the future?
23:17.54wyoungor is there a time out or something I can set which re-registers the peer if it can't be contacted in x amount of time?
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23:56.07[TK]D-Fenderno
23:56.13[TK]D-Fenderthat idea is backwards
23:56.24[TK]D-Fenderthe PEER registers
23:56.28[TK]D-Fenderit isn't the server doing it

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