| 00:18.37 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:18.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:56.15 | Labyrinth00 | Hi, I am trying to use pjsip with asterisk 15 but I can't get it to work I made a simple pjsip conf file and when I try to connect a softphone I'm seeing chan_sip reporting wrong password |
| 01:58.00 | Samot | What port is PJSIP bound to? |
| 01:58.54 | Samot | Sounds like your softphone is trying to connect on the chan_sip port vs. the pjsip port. |
| 01:59.22 | Labyrinth00 | I'm not sure, in the conf I have protocol=udp bind=0.0.0.0 |
| 01:59.40 | Samot | That's the ip binding.. |
| 02:01.02 | Labyrinth00 | ok, if I add port=5160 would it work? |
| 02:01.46 | Samot | Actually no |
| 02:01.54 | Samot | You need bind=0.0.0.0:port |
| 02:02.03 | Labyrinth00 | Oh alright |
| 02:02.11 | Samot | So if you're using both chan_sip and chan_pjsip, they need to listen on different ports. |
| 02:02.19 | Samot | If you don't specify it assumes 5060 |
| 02:02.24 | Samot | Both can't use 5060 |
| 02:04.38 | Labyrinth00 | It worked. Thanks |
| 02:04.54 | Samot | Np. |
| 02:12.53 | Labyrinth00 | I think I have a permission problem. logger_queue_init: Unable to create queue log: Permission denied |
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| 08:27.43 | WizJin | join #sysadmin-casual |
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| 08:58.16 | wyoung | WizJin: spam? |
| 08:58.42 | WizJin | huh ? |
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| 11:53.02 | Samot | sysadmin-casual? Is that like for sysadmin's in casual clothes? Or is that more for sysadmin's with a casual attitude towards being a sysadmin? |
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| 12:22.11 | dritskalle | probably for the "shoot from the hip and use a hammer" sys admins |
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| 13:34.18 | Dovid | Is there anyone here from Hawaii here? I need a test call to a toll free number from a mobile phone there. |
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| 16:09.48 | *** join/#asterisk CameronM (d924ef26@gateway/web/freenode/ip.217.36.239.38) |
| 16:11.19 | CameronM | Hello all, I am currently trying to move the AsteriskDB which is currently stored in a sqllite3 db file on the asterisk server itself, I am trying to move this to our highavailability db server so that it can be shared between multiple instances of asterisk, is this possible? I have been unable to find anything in the documentation detailing how something like this might be achieved |
| 16:11.36 | CameronM | thanks for any help you can give! |
| 16:12.44 | file | It's not meant to be shared |
| 16:12.57 | file | It's for local storage of persistent data for the Asterisk instance |
| 16:13.51 | CameronM | I understand that this is not the intended purpose, however we do not plan to be updating any data stored in the database via the dialplan etc, this is to push out number blacklists etc accross multiple asterisk installs |
| 16:14.27 | Samot | There is nothing in Asterisk to let you connect to a "remote" AstDB |
| 16:14.34 | Samot | So work from there. |
| 16:15.04 | Dovid | what you can do is connec to the remote machine, get all the values and then set them locally |
| 16:15.10 | Dovid | what sort of data do you want to replicate? |
| 16:15.20 | Samot | Or just use AMI and push the updates. |
| 16:15.28 | [TK]D-Fender | use a real DB for your blacklist data then |
| 16:15.33 | Samot | Or that |
| 16:16.45 | CameronM | Is there a way to link an additional external DB to asterisk and query it in the dialplan? (apologies for my lack of knowledge, I have been teaching myself how to do most things with asterisk) |
| 16:17.04 | Samot | Yes. |
| 16:17.06 | Samot | ODBC. |
| 16:17.27 | Samot | There are a few dialplan functions/apps that will do this for you. |
| 16:17.50 | *** join/#asterisk lunsim (~fff112@217.145.95.6) |
| 16:18.08 | [TK]D-Fender | And plenty of external options (ARI,AGI,SHELL(), etc) |
| 16:18.17 | [TK]D-Fender | ~book |
| 16:18.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:18.36 | lunsim | hi people |
| 16:18.48 | CameronM | I have used ODBC in the past to push CDR's to a SQL database, so doing something similar would be ideal. Can anyone reccomend any reading material which might give me some more information on this? |
| 16:18.53 | CameronM | Hi @lunsim |
| 16:18.57 | Samot | Or http://wiki.asterisk.org if you want to be up to date on what commands/apps there are. |
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| 16:19.15 | [TK]D-Fender | Thebook & wiki all have very clear sections for database interaction |
| 16:19.20 | lunsim | <PROTECTED> |
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| 16:20.11 | CameronM | What is thebook? |
| 16:20.14 | lunsim | is it possible to make an attended transfer with the current version of sipml5 or sip.js |
| 16:20.42 | Samot | lunsim: Sure, if you using the in call feature code options. |
| 16:21.11 | Samot | lunsim: Outside of that, it's up to the softphone/endpoint to deal with a Transfer/Refer request. |
| 16:21.46 | lunsim | about this one |
| 16:21.54 | lunsim | refer(target[, options]) |
| 16:21.54 | Samot | I don't know. |
| 16:22.00 | Samot | Possible. |
| 16:22.00 | lunsim | in session |
| 16:22.05 | Samot | Did you read the docs? |
| 16:22.14 | lunsim | who me ? |
| 16:22.16 | Samot | You're asking endpoint/softphone specific questions. |
| 16:22.18 | Samot | Yes, you. |
| 16:22.32 | Samot | Did you read the docs on the softphones you are asking about? |
| 16:22.42 | CameronM | Thanks for your help guys! I will investigate this further and let you all know what I come up with. |
| 16:23.06 | [TK]D-Fender | <[TK]D-Fender> ~book |
| 16:23.06 | [TK]D-Fender | <infobot> Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:23.28 | lunsim | yes i read it |
| 16:23.38 | Samot | What does it say about that setting? |
| 16:23.43 | CameronM | That's perfect thanks D-Fender |
| 16:24.05 | lunsim | but you are talking abount endpoint/softphone |
| 16:24.10 | Samot | Yes |
| 16:24.15 | lunsim | but with webrtc |
| 16:24.28 | lunsim | and sipphone based on sip.js or sipml |
| 16:24.29 | Samot | What about WebRTC? |
| 16:24.33 | Samot | K |
| 16:24.37 | Samot | So do they have TRANSFER OPTIONS? |
| 16:24.46 | Samot | Is there a button that handles the TRANSFER? |
| 16:25.00 | Samot | If not then your only choice is to setup In Call feature codes to do so in Asterisk |
| 16:25.14 | lunsim | oh |
| 16:25.17 | lunsim | yes ! |
| 16:25.18 | lunsim | Making a Blind Transfer |
| 16:25.18 | lunsim | SIP.js supports making blind transfers only. A blind transfer occurs when A causes B to create a session with C. |
| 16:25.34 | lunsim | do you read this ? |
| 16:25.38 | Samot | I did not |
| 16:25.42 | Samot | I don't use SIP.hs |
| 16:25.49 | Samot | I don't use SIP.js so I have no need to read the docs. |
| 16:26.00 | lunsim | so what u are using |
| 16:26.09 | Samot | For a softphone, Bria |
| 16:26.14 | Samot | I don't do WebRTC. |
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| 16:26.22 | Samot | Because there's no real demand. |
| 16:26.34 | lunsim | okey |
| 16:28.09 | Samot | So it seems the only way you can do an attended transfer is with the in-call features. |
| 16:28.39 | Samot | Which means they would have to dial something like *2 while on the call with the person they want to transfer, then dial the other person and then send # to transfer. |
| 16:29.23 | lunsim | ha good idea |
| 16:29.41 | lunsim | i will play with it |
| 16:29.47 | lunsim | god bless you :) |
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| 17:35.44 | igcewieling | gads I hate dealing with QoS |
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| 17:48.03 | Toerkeium | hello guys, I've this in my asterisk console: https://pastebin.com/YYFMrhSv and I'm trying to find a way to block this rejected unknow sip connections. So I ran a sniffer and got some packets, and reading them with wireshark I can't find any "forbbiden" or "reject" |
| 17:48.14 | Toerkeium | do you know what should I look for in the packet itself? |
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| 17:49.40 | [TK]D-Fender | Thre is no forbidden |
| 17:49.43 | [TK]D-Fender | the call is accepted |
| 17:49.49 | [TK]D-Fender | and that's what it's warning you about |
| 17:49.59 | [TK]D-Fender | You are ALLOWING anonymous calls in the first place |
| 17:50.43 | Toerkeium | [TK]D-Fender: it's being rejected (after being accepted) |
| 17:50.55 | [TK]D-Fender | no |
| 17:51.01 | Toerkeium | isnt it? |
| 17:51.14 | [TK]D-Fender | <PROTECTED> |
| 17:51.20 | [TK]D-Fender | there is no accept THEn reject |
| 17:51.22 | [TK]D-Fender | that is not a thing |
| 17:51.40 | Toerkeium | oh, ok.. I get it, it's being accepted and then SIP/200.63.55.210-00004d8b> Playing 'ss-noservice.ulaw' (language 'en') |
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| 17:52.05 | Toerkeium | so I can't filter it at IP layer |
| 17:52.08 | Toerkeium | ? |
| 17:52.21 | [TK]D-Fender | The SIP call is 100% accepted and treaeted normally. DIALPLAN processing is arranged to play audio (how much more accepted can you get if you're actually even sending MEDIA to them?) and then deciding to hangup |
| 17:52.32 | [TK]D-Fender | <[TK]D-Fender> You are ALLOWING anonymous calls in the first place <----------------- |
| 17:52.45 | [TK]D-Fender | <Toerkeium> so I can't filter it at IP layer <- Stop ACCEPTING unauthed calls |
| 17:53.01 | Toerkeium | how can I do that? |
| 17:55.19 | Dovid | when using media caching is asterisk supposed to clear out old files? |
| 17:56.07 | [TK]D-Fender | alwaysauthreject=yes |
| 17:56.14 | Dovid | For instance I have: https://pastebin.ca/4038620 |
| 17:57.17 | Toerkeium | thanks [TK]D-Fender ! |
| 17:58.07 | Toerkeium | [TK]D-Fender: then I'll find a "reject" or "forbbiden" in the packets, right? |
| 17:58.26 | [TK]D-Fender | it'll get challenged |
| 17:58.33 | [TK]D-Fender | And then it'll fail |
| 17:59.10 | Toerkeium | I'll test it, thank you very much [TK]D-Fender |
| 18:04.27 | Toerkeium | it's workinf perfect [TK]D-Fender thank you! |
| 18:04.47 | [TK]D-Fender | You're welcome. You should be reading the sample config. |
| 18:05.14 | Toerkeium | I did long time ago, I have to do it again! |
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| 20:14.04 | Toerkeium | is it normal that the first packets arrived to asterisk while registering a sip account, they are Unauthorized and later Registered? |
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| 20:15.14 | Samot | Yes |
| 20:19.14 | [TK]D-Fender | <[TK]D-Fender> it'll get challenged |
| 20:19.19 | [TK]D-Fender | <[TK]D-Fender> alwaysauthreject=yes |
| 20:19.22 | [TK]D-Fender | Means what it says |
| 20:19.32 | Toerkeium | Samot: do you know how can I identify, in the packet, when it's final result was unauthorized or registered? I'm trying to deny access to sip accounts which are unautorized 3 times, but as UDP is stateless I can't find a way to count how many times a sip account tried to auth |
| 20:19.45 | [TK]D-Fender | First it'll say "no, then the device should come back with auth saying "please", and then * will "OK" |
| 20:20.00 | [TK]D-Fender | Do you see the device come BACK with auth? |
| 20:20.15 | Toerkeium | yes I see it |
| 20:20.37 | Toerkeium | but I can't count the connections |
| 20:21.26 | Samot | Unless you tell it otherwise it will keep attempting to auth until it does |
| 20:22.03 | Toerkeium | I have to find another way, as packets will always contain the Unauthorized word |
| 20:22.12 | Samot | That's not how it works. |
| 20:22.23 | Samot | CSeq is what you need to look at. |
| 20:22.42 | Toerkeium | let me check |
| 20:22.52 | Samot | The first REGISTER is going to have a CSeq: 1, the 401 Unauthorized will have the same. |
| 20:23.05 | Samot | Then when the second REGISTER is sent it will have CSeq: 2 (this has the auth) |
| 20:23.21 | Samot | If it auth's correctly they will send a 200 OK back with the same CSeq. |
| 20:25.03 | Toerkeium | Samot: and if it fails? |
| 20:25.51 | Samot | Should be a 403 Forbidden |
| 20:26.11 | Samot | But I can't guarantee what they are sending back |
| 20:26.33 | Toerkeium | ah ok, I already have a rule for that, and it works |
| 20:27.10 | Toerkeium | I've added this "Unauthorized" trying to deny access to intents of "Unknown sip connections" |
| 20:27.32 | Toerkeium | and adding the Unauthorized, some users were denied |
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| 22:28.34 | chris349 | I thought there was a way, in voicemail.conf, to not play the instructions "please leave your message.... when finished...." to the caller but I cant seem to find it right now. Any clue? |
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| 22:55.30 | kunwon1 | chris349: it's in the voicemail application, `core show application voicemail` |
| 22:55.39 | kunwon1 | chris349: the flag 's' is what you want i think |
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