| 00:07.40 | *** join/#asterisk robmal (robmal@karaluch.xyz) |
| 00:19.29 | *** join/#asterisk tsal (~tsal@i59F4A683.versanet.de) |
| 00:20.51 | *** join/#asterisk tafa2 (~tafa2@t.ldn.dsrtnet.com) |
| 00:49.21 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 01:01.33 | *** join/#asterisk thansen (~thansen@192.74.130.86) |
| 02:02.48 | [TK]D-Fender | obviously something is not configured right |
| 02:02.57 | [TK]D-Fender | thansen, ^ |
| 02:23.15 | velix | Funny, after talking with the Universitx admin, where he says his PBX didn't have a problem, it really didn't have a problem anymore :D |
| 02:23.23 | velix | I think, he silently fixed it |
| 03:12.34 | thansen | [TK]D-Fender: you're solid as a rock, been getting help from your for 13 years now I think :) |
| 03:12.42 | thansen | lol |
| 03:13.44 | thansen | according to sip show settings: SIP address remapping: Enabled using externaddr |
| 03:16.27 | [TK]D-Fender | Just because 1 thing is set doesn't mean you configured it to refer to it properly |
| 03:18.02 | thansen | sure, again, the configs are copied verbatim from another install but this is another environment..so |
| 03:19.20 | thansen | I've got (and had set for many many years) nat=no on the existing host and all inbound stuff shows the public address and all outbound show the "From" as the public address as well |
| 03:19.42 | thansen | I've tried tweaking that setting, and various other in the new environment to no avail :( |
| 03:34.14 | thansen | [TK]D-Fender: any tips on where to dig in to find out why the externip is seemingly not being used? |
| 03:35.05 | [TK]D-Fender | You keep saying "but I did everything right" |
| 03:35.14 | [TK]D-Fender | And you haven't shown anything |
| 03:36.05 | thansen | I haven't said that, I've said it's 'right' in 1 setup and the new setup has copied the configs |
| 03:36.26 | thansen | with extremely minimal changes..what would you like to see or what's relevant to share? |
| 03:36.27 | [TK]D-Fender | same thing |
| 03:36.34 | [TK]D-Fender | show the failure and the configs |
| 03:38.26 | thansen | you want sip debug output, is that what you mean? |
| 03:38.47 | [TK]D-Fender | clearly the full comms of the call |
| 03:39.04 | [TK]D-Fender | From * CLI, not some external dump |
| 03:45.26 | *** join/#asterisk hfb (~hfb@cpe-172-117-13-65.socal.res.rr.com) |
| 03:45.27 | thansen | [TK]D-Fender: I've got more but maybe we can start with this: https://pastebin.com/0t0YDycb |
| 03:45.39 | [TK]D-Fender | no |
| 03:45.42 | [TK]D-Fender | full comm |
| 03:45.57 | [TK]D-Fender | I am not going to look at partial crap |
| 03:46.04 | [TK]D-Fender | that's why you can't find the problem |
| 03:47.55 | thansen | fair enough...I'm pretty sure the "To: foo@internal" is the problem (and where part of the behavior differs immediately) |
| 03:49.39 | thansen | [TK]D-Fender: cryptobin.co/v444n3y7 |
| 03:49.54 | thansen | can I PM the key? |
| 03:51.19 | [TK]D-Fender | yes |
| 03:53.30 | [TK]D-Fender | <[TK]D-Fender> show the failure and the configs |
| 03:54.13 | [TK]D-Fender | And you don't have basic verbose there so I can't prove what peers are used |
| 03:54.28 | [TK]D-Fender | Do it over again and include to configs |
| 03:56.14 | thansen | what do you want by 'basic verbose'? |
| 03:56.36 | [TK]D-Fender | core show verbose 10 |
| 03:56.43 | thansen | and do you mean sip.conf for config? |
| 03:56.48 | [TK]D-Fender | I don't see the DIAL command even. |
| 03:56.57 | [TK]D-Fender | And yes, obviously the sip config |
| 04:33.27 | thansen | [TK]D-Fender: cryptobin.co/d4e9j9v9 |
| 04:35.40 | thansen | some context, I'm dialing from a complete outside line to my provider which pushes the call down to the 'new' server which in turn is dialing out to my provider on another number which is getting pushed down to the 'old' server |
| 04:37.00 | thansen | when I answer the (I use this term lightly) forwarded call, there is no connection....if I hang up the call, the other end doesn't stop ringing (at least immediately, it does eventually obviously) |
| 04:40.00 | [TK]D-Fender | INVITE sip:8018771811@172.20.0.20:5060 SIP/2.0 |
| 04:40.15 | [TK]D-Fender | they are sending referencing your INTERNAL IP |
| 04:40.26 | [TK]D-Fender | Which they should never HAVE |
| 04:40.45 | [TK]D-Fender | You also don't have nat=yes under your [general] section |
| 04:41.07 | [TK]D-Fender | And I can't see if you screwed up where you even placed your register string |
| 04:41.43 | thansen | yeah on the invite, they certainly do not have the internal IP |
| 04:42.44 | [TK]D-Fender | [TK]D-Fender> INVITE sip:8018771811@172.20.0.20:5060 SIP/2.0 <- they didn't show your external IP there |
| 04:42.49 | [TK]D-Fender | which means they weren't told it. |
| 04:43.11 | [TK]D-Fender | they ignored that part and just sent to your source I'm betting |
| 04:43.12 | thansen | I've never had nat=yes on the 'old' server either and it's certainly behind the firewall with NAT going on |
| 04:44.31 | thansen | same general structure... FW -> NAT -> server with internal IP bound |
| 04:45.24 | thansen | I don't have as much insight into the NAT setup on the 'new' system, maybe the network admins tried to do some sort of fancy 'NAT' instead of just pure NATing |
| 04:45.57 | [TK]D-Fender | If you don't know how that FW is run who knows if it's mangling SIP with a transform option |
| 04:46.42 | [TK]D-Fender | But you seem to be coming in Vitel, and then going out, and not preventing re-invites which is another failure |
| 04:47.07 | thansen | the provider most certainly has no idea of the internal IP, and they couldn't be sending it to the source because it's originating from a completely unrelated cell phone |
| 04:47.08 | *** join/#asterisk ShellyRoll (~ShellyRol@c-67-170-30-252.hsd1.wa.comcast.net) |
| 04:47.39 | thansen | perhaps that's evidence enough of crazy stuff going on at the FW? |
| 04:49.04 | thansen | I *can* dial in fine to a different extension which puts me to music on hold and that works properly with audio/etc |
| 04:50.31 | thansen | I have no register string |
| 05:01.24 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 05:01.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 05:27.54 | *** join/#asterisk mir100 (~Vladimir@207.237.220.139) |
| 05:58.19 | *** join/#asterisk mir100 (~Vladimir@207.237.220.139) |
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| 09:06.04 | velix | "Received response: "Forbidden" from ..." grrrrrrrrr |
| 09:06.14 | velix | I hate the University's PBX. |
| 09:27.26 | velix | Can I change the playback for Authenticate() individually? |
| 09:44.49 | velix | Maybe I can change the language for a function only? |
| 09:48.03 | velix | Ah! Latest asterisk can override the prompt |
| 09:57.19 | *** join/#asterisk _ganapathi_ (~Ganapathi@2409:4072:6101:e297:35f7:5f4a:c4c:e399) |
| 09:57.50 | _ganapathi_ | Playback would support URI playback in asterisk 17 ? |
| 09:57.53 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 09:58.41 | velix | Hm. You cannot finetune Authenticate as it seems. "Password incorrect" is still the old text. |
| 10:02.05 | velix | Maybe 1234,1,Set(LANGUAGE()=FakeLanguage works?) |
| 10:11.38 | velix | nope |
| 10:16.30 | *** join/#asterisk deavmi (~quassel@165.255.253.203) |
| 10:19.48 | velix | Ah, indications.conf! |
| 10:30.39 | *** join/#asterisk Testeree (2d7014c7@45.112.20.199) |
| 10:30.59 | Testeree | is there a way to enable opus support (format_ogg_opus.so) when building asterisk itself? |
| 10:45.15 | velix | Testeree: you need to menuconfig it. |
| 10:45.31 | velix | I think, it's not activated by default |
| 10:47.35 | *** join/#asterisk deavmi (~quassel@165.255.253.203) |
| 10:57.49 | Testeree | ok, Thanks. Also I' am trying to enable binaural rendering support in bridges, but `./menuselect/menuselect --enable binaural_rendering_in_bridge_softmix menuselect.makeopts` is actually disabling it, while `make menuselect.makeopts` has it enabled by default. seems like MODULE_EXCLUDE is doing something in the Makefile. is it possible to forcibly |
| 10:57.49 | Testeree | enable |
| 10:59.11 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 10:59.56 | velix | Sorry, I cannot help that deep. But here are smart guys in the channels. Just stay a while :) |
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| 13:18.12 | *** join/#asterisk sysgrammer_moe (~sysgramme@d50-117-149-217.yt.northwestel.net) |
| 13:40.44 | _ganapathi_ | hi am using AGI using php. mostly am getting error as Invalid Command 510. Not able to understand from where this error coming. |
| 13:41.00 | _ganapathi_ | even for small verbose commands itself am getting this error. |
| 13:41.17 | _ganapathi_ | is it possible to understand more about this issue. |
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| 16:01.30 | Testeree | hi, anyone know how to enable binaural rendering feature in bridge during asterisk build? |
| 16:03.50 | *** join/#asterisk deavmi (~quassel@165.255.253.203) |
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| 16:10.53 | _ganapathi_ | sending VERBOSE "Init" from agi. it's 200 on one server. but same is getting failure on another server |
| 16:12.44 | _ganapathi_ | what would be the problem ? |
| 16:39.16 | Samot | _ganapahti_: Are you expecting a definitive answer? Because the best you're going to get is a vague one since that is how you've presented the problem. |
| 16:40.40 | *** join/#asterisk thansen (~thansen@192.74.130.86) |
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| 16:51.44 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 16:59.22 | *** join/#asterisk _ganapathi_ (~Ganapathi@2409:4072:604:d45d:2146:42b6:ab30:e08e) |
| 17:01.06 | _ganapathi_ | am sending some AGI commands using php based AGI program. But am not getting result from asterisk(16.4.0). |
| 17:01.27 | _ganapathi_ | getting result as Invalid Command mostly. |
| 17:02.00 | _ganapathi_ | asterisk-16.4.0. PAGI - marcelog PAGI. running AGI as laravel based commands. |
| 17:02.09 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 17:04.23 | _ganapathi_ | [2020-03-28 16:28:50] PAGI.DEBUG: Sending: VERBOSE "Init" 1 [] [] |
| 17:04.24 | _ganapathi_ | [2020-03-28 16:28:50] PAGI.DEBUG: Read: 510 Invalid or unknown command [] [] |
| 17:04.37 | _ganapathi_ | this is the sending request and got the response from asterisk |
| 17:05.52 | _ganapathi_ | ignore -- 1 [] [] |
| 17:10.07 | Samot | What does the agi debug output say? |
| 17:13.29 | *** join/#asterisk LemmeSeeThatAste (175f0f11@gateway/web/cgi-irc/kiwiirc.com/ip.23.95.15.17) |
| 17:17.20 | LemmeSeeThatAste | Hello. I'm running into an issue where if I attempt to send a text message to my user with MessageSend(pjsip:me1@192.168.10.3), neither of my linphone clients receive the message. I suspect this is similar to the issue I had with calling, where once I connected a Grandstream phone, dialing me1 only rang the GS phone. I was able to fix this for |
| 17:17.20 | LemmeSeeThatAste | calling by doing DIAL(${PJSIP_DIAL_CONTACTS(me1)}, but using that function doesn't work in MessageSend(). Is anyone familiar with how to get MessageSend() to message all clients logged into an extension? |
| 17:22.17 | Samot | You need to get all the contacts and loop them to send them all the message. |
| 17:22.42 | LemmeSeeThatAste | Are there any examples on how to do this? Any reference material on that would be helpful |
| 17:23.24 | Samot | https://community.asterisk.org/t/messagesend-to-all-pjsip-contacts/75485 |
| 17:23.47 | Samot | TBH, it was my second Google result after putting your question in. |
| 17:24.39 | LemmeSeeThatAste | Thanks. I hadn't used Google but a different search engine didn't return this. Will try it out |
| 17:25.04 | Samot | What, DuckDuckGo? |
| 17:25.19 | igcewieling | BING! |
| 17:25.30 | Samot | One of the worse search engines ever. |
| 17:25.34 | _ganapathi_ | <SIP/GATEWAY-00000067>AGI Tx >> |
| 17:25.34 | _ganapathi_ | <SIP/GATEWAY-00000067>AGI Rx << |
| 17:25.34 | _ganapathi_ | <SIP/GATEWAY-00000067>AGI Tx >> 510 Invalid or unknown command |
| 17:25.44 | Samot | There's no command. |
| 17:25.58 | Samot | > <SIP/GATEWAY-00000067>AGI Rx << |
| 17:26.04 | LemmeSeeThatAste | It was Searx configured to combine Bing, DDG and Yahoo |
| 17:26.05 | Samot | It received nothing. |
| 17:26.19 | _ganapathi_ | but before fwrite am logging. |
| 17:26.49 | igcewieling | Use an AGI library |
| 17:28.01 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 17:28.41 | igcewieling | If you are using an AGI library, then use it instead of writing the data directly. |
| 17:29.24 | _ganapathi_ | am using AGI library nly. marcelog PAGI library |
| 17:29.59 | igcewieling | you mentioned fwrite, that is part of the agi library? |
| 17:30.26 | _ganapathi_ | yes. i entered library part only to trace |
| 17:30.56 | Samot | OK well AGI didn't get any data |
| 17:32.03 | _ganapathi_ | it's look like sometime AGI receive extra input outside of AGI library. it's that possible ? |
| 17:32.23 | igcewieling | Maybe there is a bug in the library or you are using it wrong I use PHPAGI so I don't know the specific commands for that library |
| 17:33.21 | _ganapathi_ | i enabled debug on agi library. it's logging exactly which am passing. additionally empty line is passing to asterisk. anyway to identify from where it's passing.. |
| 17:34.26 | _ganapathi_ | just an empty line. it's mesh up with read variable/assign empty variable etc.. |
| 17:44.29 | _ganapathi_ | https://paste.linux.community/view/d4d2935a |
| 17:44.38 | _ganapathi_ | this is exactly the agi debug output. |
| 17:45.30 | [TK]D-Fender | You're spitting junk results back as commands |
| 17:45.44 | _ganapathi_ | https://paste.linux.community/view/bdcb8aed |
| 17:45.53 | _ganapathi_ | this is captured by library log. |
| 17:46.21 | _ganapathi_ | how to avoid this. am using php based AGI library under laravel framework |
| 17:46.36 | [TK]D-Fender | And you're pumping 2 empty lines extra each time |
| 17:46.42 | _ganapathi_ | https://github.com/marcelog/PAGI |
| 17:46.46 | [TK]D-Fender | Stop using it and use something else |
| 17:46.53 | [TK]D-Fender | or contact them about it |
| 17:46.58 | [TK]D-Fender | We don't maintain their code |
| 17:47.13 | [TK]D-Fender | And we're not seeing what your code is doing with it either |
| 17:47.28 | _ganapathi_ | ok. |
| 17:53.44 | _ganapathi_ | any official supported AGI library ?. |
| 17:55.14 | [TK]D-Fender | There is no such thing a "official" |
| 17:55.26 | [TK]D-Fender | PHP-AGI is the most popular and maintained however |
| 17:57.02 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:57.06 | _ganapathi_ | Ok. |
| 18:10.07 | igcewieling | phpagi is maintained?? |
| 18:11.27 | igcewieling | The last update I see for it is 2010. It still works great, but I would not call it "maintained". |
| 18:17.22 | _ganapathi_ | i think it's look like some stdin and stdout problem. |
| 18:17.40 | _ganapathi_ | wrongly written and captured |
| 18:28.13 | velix | Hi. Just wanted to inform you: Set(CHANNEL(language)=FakeLanguage) worked! I now can change the messages for Authenticate from the dialplan. |
| 18:39.26 | Samot | Why didn't you just use the prompt option? |
| 18:40.50 | velix | You can only change the prompt. |
| 18:40.56 | velix | Not the "pin is incorrect" message. |
| 18:41.21 | velix | Set(CHANNEL(language)) is damn nice, since I can switch languages by user entry now. |
| 18:41.24 | [TK]D-Fender | Or you could just do your own prompting and dialplan not using that app at all |
| 18:41.29 | velix | 1 for English, 2 for German etc. |
| 18:42.05 | velix | [TK]D-Fender: Sure, but this gives me flexibility to change the waveforms quickly without changing the diaplan. |
| 18:42.07 | velix | dialplan* |
| 18:42.23 | velix | Nah, forget that. names need to be equal ;) |
| 18:42.33 | *** join/#asterisk driz (~driz@joinsg.net) |
| 18:43.24 | Samot | So why are you setting a fake language? |
| 18:46.45 | velix | 1 sec, I'ver ordert stuff to drink |
| 18:47.08 | velix | Okay, I've got multiple authentications in multiple extensions and they have different texts. |
| 18:47.28 | velix | [TK]D-Fender way might be cleaner. |
| 18:47.42 | velix | Then I need to playback() |
| 18:47.44 | Samot | So why are you setting a fake language? |
| 18:48.01 | [TK]D-Fender | instead of just making a basic input dialplan for what is effectively custom |
| 18:48.02 | velix | Samot: I answered this already. |
| 18:48.29 | velix | Authenticate() only allows to change the prompt. |
| 18:48.32 | Samot | So you have different languages? |
| 18:48.37 | Samot | Or just German? |
| 18:48.53 | Samot | I didn't ask about the authenticate prompt |
| 18:48.56 | velix | Samot: No, I've got German as basic language, but I've got multiple prompts and messages for Authenticate(). |
| 18:49.05 | Samot | Sigh. |
| 18:49.13 | velix | Samot, I've got multiple prompts and messages for Authenticate()! |
| 18:49.28 | velix | You cannot make Authenticate() use another "pin is wrong" waveform without changing language. |
| 18:49.32 | Samot | Set(CHANNEL(language)=FakeLanguage) <-- Why are you doing THIS? |
| 18:49.37 | velix | sigh |
| 18:49.44 | Samot | Why isn't it de? |
| 18:49.46 | Samot | Or en |
| 18:50.02 | [TK]D-Fender | becuse he's using it to link to FAKE PROMPTS |
| 18:50.07 | Samot | or any other actual language that you can put the prompt files in |
| 18:50.17 | Samot | OK so he's just making a directory. |
| 18:50.20 | Samot | Got it. |
| 18:50.21 | velix | Sure. |
| 18:50.23 | Samot | I don't do that. |
| 18:50.39 | velix | That was the easiest solution for me. |
| 18:51.11 | velix | But I'll have a look at prompts. |
| 18:52.02 | velix | eeeh at basic inputs, sorry. |
| 18:52.44 | *** join/#asterisk Testeree (2d7014c7@45.112.20.199) |
| 18:53.01 | velix | Ah, I remember why I used Authenticate(). It allows me to read passwords from database and password files easily. |
| 18:54.59 | [TK]D-Fender | Also acheived with 1 line of dialplan. |
| 18:55.32 | velix | [TK]D-Fender: Could you hint me here? |
| 18:55.50 | velix | Authenticate() looks up a MD5 hashed password from a password file. |
| 18:55.55 | velix | That sounds complicated. |
| 18:55.56 | [TK]D-Fender | "core show cuntion SHELL" |
| 18:56.19 | [TK]D-Fender | <PROTECTED> |
| 18:56.25 | [TK]D-Fender | can't type yet.... |
| 18:56.29 | velix | and then do it in python/perl/bash ? |
| 18:57.06 | Samot | ${(DB_EXISTS(/AUTH/pins/${PIN})} |
| 18:57.15 | Samot | True/false |
| 18:57.20 | Samot | Do something based on that. |
| 18:58.56 | [TK]D-Fender | I'm betting he won't be putting it in AstDB |
| 18:59.20 | velix | Why? I could. password file was easier to start with. |
| 19:00.31 | [TK]D-Fender | This is a simple grep |
| 19:01.38 | velix | Yeah, true. |
| 19:01.49 | velix | ;) |
| 19:19.11 | *** join/#asterisk setham (~textual@unaffiliated/setham) |
| 19:38.53 | velix | [TK]D-Fender & Samot: Got it working! Thanksa lot. |
| 19:39.06 | velix | Set(Attempts=${MATH(${Attempts}+1,i)}) <-- love this |
| 19:44.15 | [TK]D-Fender | Set(Attempts=$[{Attempts}+1]) |
| 19:44.31 | [TK]D-Fender | way to overcomplicate that... |
| 19:46.30 | velix | [TK]D-Fender: Sorry :( |
| 19:47.37 | velix | But you're right. That's more nice than my old one. |
| 19:47.41 | velix | I'm even more flexible now. |
| 20:02.28 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 20:11.14 | velix | Just tested with 10 people. Stable as hell. |
| 20:32.48 | velix | I'm tesing some commercial TTS right now. Some sound aweful. |
| 20:32.55 | velix | I really think of using Siri :D |
| 20:35.54 | sibiria | google's WaveNet TTS is free to use for small amounts |
| 20:36.08 | sibiria | caveat: you give google full insight into your communications |
| 20:36.58 | sibiria | iirc amazon's TTS is also free to use within limits |
| 20:40.08 | velix | Thanks. I'll check how their German is ;) |
| 20:40.26 | velix | The commercial solutions have phonemic editors, which is pretty nice. |
| 20:40.34 | velix | The TTS in public transport often is aweful. |
| 20:41.27 | sibiria | phonemic editor in amazon's, too |
| 20:41.31 | sibiria | can't recall about google's |
| 20:45.46 | igcewieling | The problem with most google products I've investigated, like hosted VMs, TTS or STT is the site is so complicated and confusing that it takes forever to accomplish anything. |
| 20:46.50 | igcewieling | Several sites will to one-time TTS for free, which can be useful for creating custom prompts. |
| 21:03.41 | velix | That sounds illegal to me: https://ttstool.com/ |
| 21:04.04 | velix | No Access to Amazon & Microsoft non-free, but no terms of use etc. |
| 21:04.09 | velix | ew.. |
| 21:04.12 | velix | Access to Amazon & Microsoft non-free, but no terms of use etc. |
| 21:08.18 | velix | My god. I just have found a TTS, which sounds really really perfect (at least for the German language): https://responsivevoice.org/ |
| 21:10.42 | Nivex | I use a custom script with a 32KHz Festival voice. It's decent, and doesn't depend on an external service. https://octodon.social/@nivex/101828980536124369 https://gist.github.com/nivex/d03754dc5bad060fc64c162b7954a7ee |
| 21:12.21 | Nivex | Doesn't help with German though :( |
| 21:25.29 | velix | Ah! Amazon bought Ivona! |
| 21:25.36 | velix | I know Ivona from 20 years ago |
| 21:51.44 | *** part/#asterisk _ganapathi_ (~Ganapathi@2409:4072:604:d45d:2146:42b6:ab30:e08e) |
| 23:23.28 | *** join/#asterisk in8 (~in8sworld@ool-182d962b.dyn.optonline.net) |