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| 02:46.14 | hans109h | After upgrading to Ubuntu 19.04 from 18.10 I cannot start asterisk, which dies with a segfault. Core dump files: https://termbin.com/lv0t (full) https://termbin.com/go5a (thread1) https://termbin.com/icd2 (locks) |
| 02:46.46 | hans109h | Any help would be appreciated |
| 02:51.36 | Samot | Sure, please provide actual details so help can be provided. |
| 02:52.18 | Samot | Output is great but what version of Asterisk? |
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| 02:55.39 | Samot | And everything else.... |
| 02:56.08 | Samot | hans109h_: What version of Asterisk and what version of everything else? |
| 02:56.40 | Samot | You've upgraded to a 7 days old release of a new OS version that came with a new kernel update. |
| 02:56.41 | hans109h | version is Asterisk 16.2.1~dfsg-1 |
| 02:57.22 | Samot | What new versions of things are on there? |
| 02:57.32 | hans109h | on kernel 5.0.0-13 |
| 02:57.39 | Samot | Yes, which is new. |
| 02:58.09 | Samot | I'm getting at there maybe changes to libraries, etc that Asterisk isn't set to use. |
| 02:58.14 | Samot | Or expect, because they're new. |
| 02:58.29 | hans109h | I understand. |
| 02:58.57 | Samot | You're basically the first person, that I've seen in here or on the forums that has done this. |
| 02:59.00 | Samot | Soo.... |
| 03:00.50 | Samot | Is that Asterisk package from Disco Dingo? |
| 03:00.55 | Samot | Or the previous? |
| 03:02.00 | hans109h | It's from the Disco Dingo repo |
| 03:02.40 | hans109h | Perhaps it might be helpful to reboot to the kernel I was running with 18.10 |
| 03:02.56 | Samot | Did you do that and it fixed it? |
| 03:04.01 | hans109h | I haven't tried yet, was just thinking out loud |
| 03:04.31 | hans109h | dmesg contains a lot of lines like traps:"asterisk[3361] general protection fault ip:558fc33fe230 sp:7fff1fd20020 error:0 in asterisk[558fc32ca000+1ae000]" |
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| 03:05.29 | hans109h | I'm going to reboot to previous kernel and will report back momentarily |
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| 03:14.40 | hans109h_ | Using a previous kernel had the same results |
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| 14:00.58 | Jonsku | Hello good people! |
| 14:05.14 | Jonsku | I'm working on a project using AMI to originate a call to a phone on the PSTN via a SIP Trunk and using the JACK application in the dialplan to interact with the input and output audio stream. |
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| 14:10.21 | Jonsku | This is working fine, but now I would like to detect when all the audio my application is generating for output was delivered to the callee. My requirement is to be able to hang up the call as soon as the audio has been delivered, because at the moment I have no way of knowing exactly when to hang up the call (I could use a timer, but I'm hoping there is a better solution). |
| 14:14.24 | Jonsku | I've been trying the TALK_DETECT function, but I can not figure out how to make it work together with the JACK application, and by that I mean that I only get 'talking' and 'silence' events for the incoming audio (the one coming from the phone), not for the outgoing audio coming from JACK AUDIO. |
| 14:16.20 | file | TALK_DETECT isn't meant for that direction, it's meant for audio from the channel to know when they're speaking |
| 14:17.48 | Jonsku | Ok, thanks for clearing this up. So I can rule the TALK_DETECT solution out. |
| 14:20.36 | Jonsku | However, would there be a way to have a local channel 'listen on' the output audio of the call, and perhaps apply the TALK_DETECT function there? |
| 14:23.50 | file | ChanSpy can be used to spy on a channel, when used with a Local channel maybe |
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| 14:29.46 | sibiria | Jonsku: if you're not expecting to see latencies between you and the nearest point to reach 1 second, why not just go by that measure or something smaller? |
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| 14:30.30 | sibiria | you could monitor latency via QoS, but i wonder if it's not an unnecessarily complicated solution |
| 14:30.48 | sibiria | play your audio, wait 500ms, hang up |
| 14:32.26 | sibiria | you know.. K.I.S.S |
| 14:33.02 | file | +1 |
| 14:34.43 | Samot | You wanted the best, you got the best. |
| 14:34.50 | Samot | Oh wait, the other KISS |
| 14:35.02 | Jonsku | @sibiria: Yes, starting a timer to hangup the call this is probably what I will end up doing. It just feels a bit brittle, but hopefully in practice it will work OK. |
| 14:36.04 | sibiria | what feels brittle is relying on analysing the audio stream to know when all audio is played |
| 14:36.28 | sibiria | what if the message has a pause? what if you want to pause between two audio files etc. |
| 14:36.51 | sibiria | whatever is playing the audio is what should signal to hang up |
| 14:39.43 | Jonsku | Actually, I only need to do this for the last audio stream I'm sending, this is not needed otherwise. |
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| 14:43.18 | Jonsku | "whatever is playing the audio is what should signal to hang up" > true, but only for the sake of the argument, wouldn't it be better for the signal to be triggered by the end of the audio pipeline rather than its source? |
| 14:43.32 | sibiria | no. |
| 14:43.52 | sibiria | that obviously limits the nature of the audio |
| 14:44.14 | sibiria | ie. you can't have a pause longer than N milliseconds or the call will hang up |
| 14:46.50 | Jonsku | And if, as in my case, it can be guaranteed that there will not be a silence longer than N? |
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| 14:49.55 | Jonsku | Anyway, thanks for your input and your time, very helpful! |
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| 15:23.52 | Kobaz | mmmm, yealink question.... have a three way conference... hang up... all calls drop |
| 15:24.27 | Samot | Are you doing it locally on the Yealink? |
| 15:24.31 | Kobaz | is there an option to have it behave like a polycom, and continue the call as a two party call after the third (bridging) party hangs up |
| 15:24.35 | Kobaz | locally |
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| 15:36.33 | [TK]D-Fender | Kobaz, throw out the Yealink and get a Polycom :p |
| 15:37.26 | Samot | And also get the right model. Not all support what you're referring to. |
| 15:37.36 | Samot | Lower end models will do what you just described. |
| 15:38.18 | Samot | But I'm not sure if the Yealink's can do a blind transfer/refer like that |
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| 15:43.04 | Kobaz | [TK]D-Fender: getting polycoms will bump the price on this job by 40k |
| 15:46.04 | [TK]D-Fender | Then go pay $20k to Yealink to have them add it to their firmware ;) |
| 15:54.38 | Samot | Or program the Yealink to use the attended transfer feature code options. |
| 15:55.09 | Samot | That allows you to 3-way the call and then join the other two parties together when the middle man hangs up. |
| 15:55.51 | Samot | The work around is to not do it locally if the intent is to join to random parties at the end. |
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| 16:14.01 | Kobaz | this is with attended transfer turned on |
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| 16:44.20 | file | hans109h: there's an old Asterisk module in the modules directory (if you haven't figured that out yet) |
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| 16:48.05 | hans109h | file, I haven't figured it out yet....are you saying that I have something that should be removed or do I need to replace it with something? |
| 16:48.35 | file | you have a module in the modules directory (by default /usr/lib/asterisk/modules) that is old and not supported on the version of Asterisk installed |
| 16:49.40 | file | the symbols are incomplete, so I can't say what one |
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| 16:49.50 | hans109h | Hmmmm...perhaps app_notify |
| 16:50.05 | file | that does not exist in the tree, so possibly |
| 16:51.03 | hans109h | I got 5 modules from January 2019, the rest are from March 2019, and then app_notify so I will start there. |
| 16:51.17 | hans109h | Thank you kindly for pointing that out. |
| 16:52.27 | file | no prob |
| 16:52.39 | hans109h | file! You solved it! |
| 16:52.54 | file | someone in FreePBX land experienced the same thing, so it was on my memory |
| 16:53.15 | hans109h | cool. |
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| 17:51.54 | life_of_e_ | playing around with calendar integration but it doesn't appear that asterisk connects to my calendar server (Radicale caldav) |
| 17:52.13 | life_of_e_ | no calendars show up in calendar show calendars |
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| 18:17.06 | Samot | Wrong channel. |
| 18:17.37 | life_of_e_ | why is it the wrong channel? It's asterisk that's not connecting |
| 18:18.08 | Samot | Sorry used to FreePBX users asking things here. |
| 18:21.38 | life_of_e_ | No worries. I'm just not quite sure what I'm missing since it seems calendar integration is relatively straightforward. Add entry to calendar.conf, reload res_calendar.so, done. |
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| 18:22.46 | life_of_e_ | I'm not even trying to use it for notifications, I'm actually wanting to write to the calendar |
| 18:23.42 | life_of_e_ | The log on the calendar server never shows a connection attempt |
| 18:23.57 | life_of_e_ | But asterisk never shows an error after module reload either |
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| 18:53.09 | life_of_e_ | ah, found it, the caldav version of the module never compiled |
| 18:53.24 | life_of_e_ | but I can't seem to meet the dependencies for it, oh well |
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| 19:39.43 | life_of_e | What's the IXJUSER dependency? |
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| 22:33.47 | life_of_e | Ok, so the h extension in a context is triggered when the channel is hung up. Is there a similar |
| 22:34.13 | life_of_e | Is there a similar way to use something like s? Will that get triggered if a call comes in and then transition to the dialed extension? |
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