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00:36.29 | JavierLazo | hola alguien aqui que hable espanol??? |
00:36.49 | JavierLazo | pregunta si hay alguien que hable espanol y lo pueda ayudar |
00:37.43 | hans-christian | Quiero que esto no comprar la alfombra :> |
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00:38.31 | JavierLazo | Hello, 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 |
00:39.35 | hans-christian | I add also some problem. And got directly some help :> |
00:39.39 | hans-christian | erm "had" |
00:39.52 | hans-christian | ask you questions and do ask to ask |
00:40.00 | JavierLazo | someone can help me with my settings sip.conf and my extensions.conf |
00:40.22 | JavierLazo | I have already registered the trunking |
00:40.36 | JavierLazo | but when I try to dial calls will not let me |
00:40.44 | JavierLazo | example |
00:40.45 | JavierLazo | [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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00:52.31 | JavierLazo | thanks |
00:52.49 | JavierLazo | robink_: |
00:52.54 | JavierLazo | hello |
00:53.23 | JavierLazo | i needdddddd heelllllppppppp |
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00:55.59 | JavierLazo | hello |
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00:57.56 | JavierLazo | raspberrypifan: |
00:58.02 | JavierLazo | hi |
00:58.07 | raspberrypifan | hi |
00:59.42 | JavierLazo | raspberrypifan: |
00:59.54 | JavierLazo | I need help |
01:00.26 | JavierLazo | I spent 4 days trying to configure my asterisk, and have failed |
01:00.36 | raspberrypifan | what happeend |
01:00.45 | JavierLazo | you can help me please |
01:00.47 | JavierLazo | ok |
01:01.26 | JavierLazo | and record the trunk but now I need that incoming calls to my softswitch, who come from another server out my asterisk |
01:01.27 | raspberrypifan | what |
01:01.48 | JavierLazo | You understand |
01:01.50 | JavierLazo | ok |
01:01.55 | JavierLazo | i have mi asterisk ready |
01:02.02 | JavierLazo | the sip trunk |
01:02.04 | raspberrypifan | si |
01:02.10 | raspberrypifan | cuenta el problema de una vez |
01:02.15 | JavierLazo | the sip trunk |
01:02.20 | JavierLazo | hablas espanol |
01:02.21 | JavierLazo | ok |
01:02.35 | JavierLazo | tengo mi asterisk registrado con la troncal sin problema |
01:02.37 | JavierLazo | localhost*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> |
01:03.06 | JavierLazo | ahora lo que necesito es que desde mi softswitch yo pongo como end point carriers mi asterisk o sea el ip puerto 5060 |
01:03.14 | JavierLazo | las llamadas cuando entran a mi asterisk me da esto |
01:03.42 | JavierLazo | [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'. |
01:03.49 | JavierLazo | y no conecta la llamada al destino |
01:04.10 | JavierLazo | raspberrypifan: tienes idea como ayudarme |
01:04.12 | JavierLazo | ? |
01:04.33 | JavierLazo | es problemas en mis config que no son buenas |
01:04.37 | JavierLazo | y necesito ayuda con eso |
01:04.43 | raspberrypifan | pero copia la configuracion sample |
01:04.45 | raspberrypifan | y ahi le cambias |
01:04.49 | raspberrypifan | el problemas es el contexto |
01:05.10 | JavierLazo | llevo 4 dias y no logro que funcione |
01:06.19 | raspberrypifan | coppia el sip ejemplo |
01:06.20 | raspberrypifan | y ahi busca |
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03:51.34 | raspberrypifan | [TK]D-Fender: hello friend |
03:58.12 | raspberrypifan | i cant get yate to register with my asterisk vm install |
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07:54.22 | tparcina | How to specify default time zone in voicemail.conf? |
07:54.43 | tparcina | So that I don't have to specify it for every account. |
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10:52.01 | davlefou | tparcina, Bonjour, il faut configure des groupes selon les zones choisis. |
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12:08.33 | amonk | tparcina: [zonemessages] |
12:08.33 | amonk | eastern=America/New_York|'vm-received' Q 'digits/at' IMp |
12:08.33 | amonk | central=America/Chicago|'vm-received' Q 'digits/at' IMp |
12:08.33 | amonk | central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' |
12:08.33 | amonk | military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' |
12:08.36 | amonk | european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM |
12:09.02 | amonk | see [zonemessages] in /usr/local/share/examples/asterisk/default/voicemail.conf or whatev on your system |
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12:11.11 | tparcina | amonk: TY |
12:11.22 | amonk | sure |
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13:31.46 | Rasputin3711 | Hello |
13:32.05 | Rasputin3711 | What is the best version Asterisk for production system ~50 users? |
13:32.42 | [TK]D-Fender | Any that is current and supported |
13:32.48 | [TK]D-Fender | See !topic |
13:32.55 | Rasputin3711 | 1.8.xx or 13? |
13:32.59 | [TK]D-Fender | 13 |
13:33.01 | [TK]D-Fender | 1.8 is DEAD |
13:33.17 | [TK]D-Fender | Topic 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 |
13:33.48 | Rasputin3711 | I know, but i see a lot of topic then people don't recomende 13.8. |
13:34.41 | [TK]D-Fender | I'd be interested to see who, where, and why... |
13:35.00 | Chainsaw | 11 is the lowest you can go, but for a new install, I'd go with 13. |
13:35.04 | [TK]D-Fender | The only supported branches are 11 & 13. |
13:35.11 | Chainsaw | It's where the development effort is going. |
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13:35.15 | [TK]D-Fender | And 11 Won't be supported that much longer |
13:35.45 | Rasputin3711 | Do you have systems on 13 ver with more than 100 users? |
13:35.49 | file | yeah, I'd be interesting in seeing the reasons why people aren't using 13 |
13:35.50 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
13:35.54 | [TK]D-Fender | 100 users is nothing |
13:36.24 | [TK]D-Fender | they've pushed upwards of 10000 calls through a single server ( under controlled circumstances) |
13:36.42 | Rasputin3711 | What OS? |
13:36.45 | [TK]D-Fender | Linux |
13:36.50 | Rasputin3711 | Distro? |
13:36.58 | [TK]D-Fender | Doesn't matter |
13:37.03 | [TK]D-Fender | No magical difference |
13:37.39 | [TK]D-Fender | Asterisk doesn't care. It is either built properly and works, or was not and did not. |
13:37.54 | Chainsaw | And an element of personal preference always creeps into distro choices. |
13:37.57 | Rasputin3711 | Do i need update my system 13.7 -> 13.8.2 or it's not necessary? |
13:38.10 | [TK]D-Fender | 13.8.2 came out for a reason |
13:38.16 | Chainsaw | Rasputin3711: Always go for the most recent version. Bugs get fixed all the time. |
13:38.33 | [TK]D-Fender | There are obviously some mix of bug & security fixes, and possibly new features |
13:38.48 | Gugge | And new bugs :) |
13:39.03 | Rasputin3711 | Where can i find bugs for certain version for example(13.8.2)? |
13:39.05 | [TK]D-Fender | I want only the newest, most fashionable bugs |
13:39.09 | file | we've actually been pretty good |
13:39.11 | [TK]D-Fender | in the changelog |
13:39.22 | [TK]D-Fender | It's included in the tarball and on the websitte |
13:39.33 | Rasputin3711 | After release |
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13:40.16 | Chainsaw | file: My downstream patchset isn't growing, which is a really good sign. |
13:40.57 | file | our CI infrastructure is finding stuff before merging, which is really |
13:41.21 | [TK]D-Fender | There is a bug tracker as linked off the main site |
13:41.34 | file | good |
13:41.36 | file | missed a word |
13:41.38 | [TK]D-Fender | For current issues being examined |
13:42.21 | [TK]D-Fender | Typically new bugs aren't all completely new, they are just older issues being discovered. |
13:42.52 | [TK]D-Fender | So 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-Fender | So unless there is a very specific issue that doesn't hit a slightly older release then there is no sense in starting backwards |
13:43.26 | Rasputin3711 | 13.6.0 has good bug with sip and i don't want to get it another time. |
13:43.36 | [TK]D-Fender | Because that ignores things that have actually been fixed |
13:43.54 | [TK]D-Fender | and was that bug fixed? |
13:44.08 | [TK]D-Fender | Do you see it re-issued? |
13:44.28 | Rasputin3711 | It was closed in the next version 13.7. |
13:44.38 | Rasputin3711 | But it was not fun in production. |
13:45.24 | [TK]D-Fender | was that bug only in 13.6.0? Did it list as applying to earlier versions? |
13:47.20 | Rasputin3711 | Yes, https://issues.asterisk.org/jira/browse/ASTERISK-25476 |
13:48.24 | [TK]D-Fender | Think I remember that one coming up a few times |
13:48.38 | [TK]D-Fender | Was a bigger one.... |
13:48.57 | [TK]D-Fender | So keep an eye on the tracker and jsut start witht he latests |
13:49.19 | Rasputin3711 | What is your rule how to update to new version? |
13:50.39 | [TK]D-Fender | Test 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-Fender | Basically : common sense. |
13:50.58 | [TK]D-Fender | Don't be gun-shy about it |
13:51.13 | [TK]D-Fender | Seeing the changelog to see what's fixed is always a good idea as well |
13:52.36 | file | from 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 |
13:56.16 | Rasputin3711 | Many thanks! |
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17:15.11 | drale2k | Hey 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) |
17:15.43 | drale2k | Can i now have the iPhone app configured externally (using my public server IP) and use it or is the ping / delay too high ? |
17:15.54 | drale2k | or is asterisk fine with 30-50 ms ? |
17:18.33 | newtonr | it'll probably work. it will however be highly variable in call quality |
17:19.53 | drale2k | newtonr: hm what is the better solution? I would like something that does not go over landline |
17:20.36 | drmessano | I use an iOS softphone without too many issues |
17:20.50 | drmessano | You hit a bad spot and obviously youre stuck |
17:21.08 | newtonr | Yeah I mean if you have a strong data connection it can work just fine |
17:21.13 | drale2k | drmessano: is it just configured with the Asterisk Server IP ? |
17:21.22 | drmessano | It wont be _as_ reliable as your native calls on your phone |
17:21.25 | drmessano | Yep |
17:21.35 | drale2k | useable is good enough for me |
17:21.45 | drale2k | like hear clearly |
17:21.52 | drmessano | Yes |
17:22.05 | drmessano | Its actually been a lifesaver |
17:22.20 | newtonr | be 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. |
17:22.31 | drale2k | newtonr: yup sure |
17:22.35 | drmessano | I have a few locations where I have no cell service, but I do have Wifi |
17:22.42 | drmessano | So that works |
17:22.50 | drmessano | and 100% what newtonr said |
17:22.52 | drale2k | drmessano: ye, does it work well with 3G ? |
17:23.03 | drmessano | When you say "3G" |
17:23.11 | drmessano | Do you mean "3G" or are you being generic? |
17:23.17 | drale2k | 3G the phone network |
17:23.24 | drale2k | like no WiFi but 3G |
17:23.25 | drmessano | Thats not what I am asking |
17:23.48 | drmessano | Most places are 4G/LTE |
17:24.04 | drmessano | Are you specifically talking about 3rd generation networks? |
17:24.16 | drale2k | i am asking 3G, not 4. the thing before 4G / LTE |
17:24.23 | drmessano | Ah |
17:24.29 | drale2k | because i expect it to work with LTE |
17:24.49 | drmessano | 3G works.. I would say maybe 20% less reliable than 4G |
17:24.56 | drmessano | The 3G networks sucked |
17:25.01 | drale2k | by reliable you mean quality or break ups |
17:25.09 | drale2k | like disconnects etc |
17:26.12 | drmessano | A Call dropping is not a quality issue, its a capacity issue |
17:26.21 | drmessano | and thats obviosuly not an issue |
17:26.29 | drmessano | and thats obviosuly not an issue with your own server |
17:26.47 | drmessano | But on 3G you may have jitter and could lose connection |
17:27.00 | drale2k | lose connection is a call drop for me :P |
17:27.04 | drale2k | but i get what you mean |
17:27.10 | drmessano | Not necessarily |
17:27.22 | drale2k | could it just reconnect ? |
17:31.32 | *** join/#asterisk ElliotJohn (be352905@gateway/web/freenode/ip.190.53.41.5) |
17:32.39 | ElliotJohn | Hi guys, I´m kind of new in this Asterisk business...need some help in choosing a box |
17:34.36 | Synthase_ | Choosing a box? |
17:36.24 | drale2k | a phone probably |
17:39.23 | ElliotJohn | 480 concurrent calls |
17:39.53 | ElliotJohn | . 600 user sites |
17:40.21 | ElliotJohn | support is required for FXO/FXS/PRI/BRI circuits |
17:40.44 | ElliotJohn | Concurrent calls to be supported + allowance for any CODEC transcoding to be done + allowance voice recording to be done |
17:41.11 | ElliotJohn | DISC SIZE = 500GB + voicemail space + voice recording space |
17:42.15 | Synthase_ | Transcoding varies by codec, I wouldn't leave that open ended as some are more CPU intensive than others. |
17:42.38 | ElliotJohn | CDR/reporting can be handled, VM |
17:46.30 | ElliotJohn | 2 x 10/100 NICs running initially in primary/secondary HA mode |
17:49.51 | ElliotJohn | so 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 |
17:50.41 | ElliotJohn | what brand/model do you guys recomend ?? |
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18:57.51 | drale2k | i 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 |
18:58.00 | drale2k | it works fine internally |
18:58.14 | drale2k | its just for testing so i am using my public home IP |
18:58.27 | drale2k | asterisk server runs on my laptop |
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19:13.02 | wyoung | Bacon 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!!!! |
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19:16.52 | Nugget | wyoung: https://www.youtube.com/watch?v=cUYSGojUuAU |
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20:44.55 | CryptoManiac | Where's good old [TK]D-Fender ? |
20:45.15 | CryptoManiac | He's always here, weird not to see him here |
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20:54.57 | wyoung | Nugget: :D |
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21:48.26 | [TK]D-Fender | CryptoManiac, Was in-transit home |
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22:15.06 | CryptoManiac | [TK]D-Fender: Thought so. (That you will be away for a minimal time). |
22:15.55 | CryptoManiac | Someone else needed some help with a jitter buffering issue so I referred him to you |
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23:16.32 | wyoung | so I get a subscriber absent error when dialling out. I restart asterisk and all is good :\ |
23:16.51 | wyoung | Any way to tell asterisk to do a restart if this happens in the future? |
23:17.54 | wyoung | or 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-Fender | no |
23:56.13 | [TK]D-Fender | that idea is backwards |
23:56.24 | [TK]D-Fender | the PEER registers |
23:56.28 | [TK]D-Fender | it isn't the server doing it |