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| 07:19.33 | Willi_ | Guten Morgen zusammen! |
| 07:19.49 | Samot | English only |
| 07:20.03 | Willi_ | Sorry, good morning @all |
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| 07:40.26 | Willi_ | I have a little question: we have in our house a lot of analog phones (>10). When we switch to an Asterisk based Phone system, can we throw all phones into the trash? I read about ATA Adapts and Gateways, but did we need one for each phone? |
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| 09:22.57 | Willi_ | No one? |
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| 14:06.14 | sakhi | Hello, any recommendations for an open source sms gateway, I've played with Kannel and Jasmin. |
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| 14:16.28 | Download-Fritz | hey :) Does anyone know whether there is a way to disable expiration with PJSIP (not quite sure whether the SIP protocol even allows that)? I don't actually need the SIP registration to stay active, though I need the AOR to remain in the database (via either 'database' or 'pjsip show aor', I don't care) as I'm going to use push notifications for incoming calls |
| 14:17.39 | Download-Fritz | ideally, the only times my client will connect to Asterisk is after a push notification was received or when the Push Notification token changes. When both cases don't hit in a while, even with the greatest expiration time, it would disappear. I quickly checked the source and it seems like '0' is reserved for deletion |
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| 15:10.25 | *** join/#asterisk stefan27 (~stefan27@static-212-247-4-149.cust.tele2.se) |
| 15:11.25 | stefan27 | Hmm, is there any SIP service you can call, that analyzes audio quality metrics of the remote stream? |
| 15:12.14 | stefan27 | e.g. you call "sip:boo@bla.com", it analyzes your stream and sends you feedback somehow |
| 15:12.38 | stefan27 | would be nice to have some kind of automated sound quality test |
| 15:14.25 | stefan27 | something like chrome://webrtc-internals for webrtc calls, but instead something in the middle that is not client-software dependent |
| 15:15.32 | stefan27 | I know "audio quality" is vague, there's probably a whole science dedicated to different aspects of it (noice supression, echo cancellation, absence of noticable jitter, ...) |
| 15:15.47 | Samot | You need something in the path. |
| 15:16.15 | Samot | The Chrome plugin makes Chrome the one doing it. |
| 15:19.07 | Samot | But there's nothing in Asterisk to do this. |
| 15:19.13 | Worldexe | im using voipmonitor for my Asterisk installation; not exactly what you want, but still |
| 15:19.30 | Samot | Ick. |
| 15:20.22 | Samot | The ironic thing about VoIPMonitor, is that when people complain their server is overload or has audio quality issues... |
| 15:20.22 | stefan27 | anything that might help detect, troubleshoot, prevent or understand audio quality problems |
| 15:20.40 | Samot | They turn off VoIPMonitor and the issues go away. |
| 15:20.41 | stefan27 | I'll check it out thanks |
| 15:20.46 | Samot | I wouldn't. |
| 15:20.51 | Samot | It's a resource hog |
| 15:21.00 | stefan27 | For troubleshooting, not necessarily in production |
| 15:21.00 | Samot | It causes the problems it is meant to detect in cases. |
| 15:21.03 | Worldexe | yeah, you should be careful |
| 15:21.21 | Samot | If your monitoring software causes the issues it alerts for you.. |
| 15:21.24 | Samot | You shouldn't use it. |
| 15:21.50 | stefan27 | on the quantum level you cannot observe anything without affecting it :) |
| 15:21.57 | Samot | OK |
| 15:22.08 | Samot | Why do you need this? |
| 15:22.10 | stefan27 | on the software level you just hope the effects is small enough |
| 15:22.15 | Samot | Are you having audio quality issues? |
| 15:22.27 | stefan27 | nah |
| 15:22.36 | stefan27 | some users were |
| 15:22.45 | stefan27 | If it ever happens again I wanna be prepared |
| 15:22.56 | Samot | Where are you going to run this |
| 15:23.08 | stefan27 | I don't know |
| 15:23.16 | Worldexe | and how many parallel calls with rtp do you have? |
| 15:23.30 | Samot | vs calls with no RTP? |
| 15:24.11 | stefan27 | I don't know have data for the environment from the supposed issue, I wanna get an idea what kind of data to collect next time it happens |
| 15:24.19 | Worldexe | also, voipmonitor can work being fed port-mirrored traffic; so it will not affect Ast in any cases |
| 15:24.23 | stefan27 | e.g. server load, network load, client stuff |
| 15:24.33 | Samot | What kind of issues where they having? |
| 15:25.49 | stefan27 | they thought the sound in their softphone sounded "bad" (not more specific than that) and that they had better experience with another PBX, another internet telephony service provider and another third party softphone |
| 15:26.06 | stefan27 | A) I don't know if their testimony is true, B) I don't know which component was responsible for it if it was true |
| 15:26.08 | Samot | Where is the softphone? |
| 15:26.24 | Samot | On a PC or on a mobile device? |
| 15:26.32 | stefan27 | on a PC |
| 15:26.35 | Samot | OK |
| 15:26.41 | Samot | So 1) A lot of softphones are shitty |
| 15:26.47 | Samot | 2) Uhm, it's the PC |
| 15:26.58 | Samot | You know what messes with calls on softphones on a PC? |
| 15:26.59 | Samot | The PC |
| 15:27.02 | Samot | It's load. |
| 15:27.07 | Samot | It's memory |
| 15:27.12 | Samot | Programs locking up |
| 15:27.14 | Samot | Lag. |
| 15:27.17 | stefan27 | It's impossible to solve *THAT* particular issue, that person may have moved to hawaii now for all I care, I just want a better world's view for next time |
| 15:27.19 | Worldexe | are the second party was also on your installation? the dab quality can arrive from beyoud your control |
| 15:27.22 | Samot | All that can impact a softphone. |
| 15:27.24 | fauxalliance | jitter and other packet loss |
| 15:27.42 | Samot | "bad" is vague. |
| 15:27.49 | stefan27 | yeah |
| 15:27.50 | Samot | Stop chasing your tail. |
| 15:27.51 | sibiria | the most obvious explanation isn't that the person calling on the softphone was doing so when the PC was heavily burdened by other software, but simply dodgy network at the time |
| 15:27.57 | Samot | Just stop. |
| 15:28.10 | Samot | <-- Been doing this for 12 years |
| 15:28.25 | Samot | 50K+ user bases |
| 15:28.29 | Samot | Worldwide. |
| 15:28.40 | Samot | I've dealt with almost every issue an end user can have |
| 15:29.05 | fauxalliance | agreed, and Samot also tends to do things 'the right way' not the easy way. |
| 15:29.13 | stefan27 | Some softphones are more resiliant to dodgy network. Since we are re-distributors of softphone we want to get statistical data of which softphones work best in which environments... And we also want to find out if changing some of the softphone settings can positively change their experience |
| 15:29.26 | Samot | My point is simple. |
| 15:29.29 | sibiria | stefan27: it's easy to measure jitter through rtpqos. i'd start there |
| 15:29.35 | stefan27 | we only have the empirical approach but it's so slow |
| 15:29.41 | Samot | You're running around trying to get all this shit in place "in case" it happens again |
| 15:29.45 | fauxalliance | usb audio hardware offloading the DSP processing, may help... even with a softphone |
| 15:29.47 | Samot | And it's *all* pointless. |
| 15:30.32 | stefan27 | DSP processing? |
| 15:30.41 | Samot | shakes his head |
| 15:30.44 | Samot | Here we go. |
| 15:31.02 | fauxalliance | that also does _nothing_ to mitigate the network issues. |
| 15:31.12 | sibiria | it may help on a 75mhz 486 :p |
| 15:31.26 | sibiria | but it's likely not the hardware said softphone was running on |
| 15:31.37 | sibiria | really, the most likely explanation was dodgy network at the time of the call |
| 15:31.50 | fauxalliance | sibiria: one never knows |
| 15:32.15 | Samot | Look.. |
| 15:32.41 | fauxalliance | _good_ call quality requires _excellent_ network quality. |
| 15:32.42 | sibiria | g.711 is an incredibly simple codec. it requires almost no processing power what so ever. same goes for g.722 and other popular choices in voip |
| 15:32.43 | Samot | The "a user reports a 'bad' call so I need to install RTP stats software to be prepared" is waaay over thinking. |
| 15:32.53 | fauxalliance | true |
| 15:33.02 | Samot | Just stop |
| 15:33.04 | Samot | Step back |
| 15:33.09 | Samot | SIMPLE things. |
| 15:33.14 | Samot | BASIC things. |
| 15:33.23 | fauxalliance | TCP/IP things |
| 15:33.32 | Samot | Stop jumping to Tier III level support without doing Tier I |
| 15:33.42 | Samot | Do triage before you do surgery. |
| 15:34.04 | sibiria | here's a simple and basic thing: rtpqos on each call is a very basic form of statistics that help tracking problems down as well as helping mitigating larger and escalating quality problems |
| 15:34.13 | Samot | Running around and setting up a bunch of shit for supporting something.. |
| 15:34.34 | Samot | THEN finding out with a bit of triage everything you did was pointless because the issue isn't what you thought. |
| 15:34.57 | Samot | Tech Support is not always complex. |
| 15:35.03 | fauxalliance | or competent |
| 15:35.06 | Samot | It can be down right plain and basic. |
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| 15:35.23 | fauxalliance | o/ |
| 15:35.33 | Samot | I've watched techs WASTE time because they skip a basic step |
| 15:35.48 | Samot | And went straight to the "Advanced" steps because, hey they're techs and that is what techs do. |
| 15:35.58 | Samot | Support issues = complex some how all the time. |
| 15:36.33 | Samot | Tech support is the biggest time and cost bleed on a provider. |
| 15:36.45 | Samot | Because it's generally done wrong and inefficient. |
| 15:38.10 | Samot | Then again, when you spend 90% of your career in a Agent:User ratio of like 1:25,000 you learn to be efficient. |
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| 18:03.51 | tehgooch | Samot: Amen. It's also important to have a good grasp on what the issue is. e.g. Bad audio. What is the nature of the bad audio? "I sound echoey" would be a completely different set of things to check (e.g. Wrong audio device selected, bad mic, etc...) than calls breaking up (Maybe packet loss, jitter buffer issues, etc...). |
| 18:04.15 | tehgooch | Of course even in that example there is overlap. |
| 18:05.21 | Samot | How about, it's a softphone running on a PC. |
| 18:05.28 | tehgooch | Haha true facts. |
| 18:05.38 | Samot | The voice network isn't the first place you look. |
| 18:05.50 | Samot | Does the issue exist on IP phones? |
| 18:06.03 | Samot | If this extension is moved to an IP phone, does it have the same issue? |
| 18:07.22 | Samot | Softphones are totally dependent on the systems they run on. If the system is overloaded, lags, buffers, whatever... |
| 18:07.30 | Samot | That is going to impact the softphone. |
| 18:07.40 | tehgooch | Higher ups recently decided selling a rebranded softphone was a good way to make revenue. Now we have to support the damn thing and I hate every ticket that comes in about it. |
| 18:07.53 | Samot | Well.. |
| 18:08.07 | Samot | Softphone on a mobile device.. |
| 18:08.15 | Samot | A little different. |
| 18:08.24 | tehgooch | That is coming. FML. |
| 18:09.16 | tehgooch | Luckily Apple is being an absolute bitch about accepting it so it's massively delayed. |
| 18:09.55 | Samot | Also, the whole "The user reported a problem. I need to look at everything on my end" just tells me you have no confidence in your own network. |
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| 18:10.41 | Samot | "Huh, 1 out of 200 users on this system is having an issue" |
| 18:10.53 | Samot | That does not equate to "Let me go over my system with a fine tooth comb" |
| 18:12.00 | tehgooch | I'm getting a little tired of the new girl escalating issues and not being able to answer what the problem actually is. She is improving though so there's that. |
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| 18:27.41 | fauxalliance | speaking of softphone on a mobile device, grandstream wave, they actually got something right. |
| 18:27.55 | fauxalliance | and the 5 user mini conference is cute |
| 18:29.12 | fauxalliance | and it doesn't shag my battery life. |
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| 18:46.03 | fauxalliance | Samael28: fix your internet |
| 18:46.20 | fauxalliance | wheeeeeeee |
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| 19:20.11 | Download-Fritz | sorry, had to leave. Did I miss any replies? :) |
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| 19:34.39 | fauxalliance | also in the world of faxing and t.38 Brother fax machines have an 'overseas' mode... works great with t.38 |
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| 19:37.18 | sebastienthiry | Hello everybody, I try to configure an asterisk ipbx but I have an error, after few seconds the phone hang up and in my log I see this "Hanging up call 24oiFLY2qtfb6mJ161GemfYjCRuzelZB - no reply to our critical packet". It is I think a nat problem. The server is behind a nat and my phones are behind an other nat, so communication has to pass by internet. |
| 19:38.43 | sebastienthiry | Anybody can help my? I have already configure this options externhost, localenet, nat=yes |
| 19:38.49 | fauxalliance | sebastienthiry: try putting the entire log into a pastebin... i'm sure someone will take a look |
| 19:39.08 | sebastienthiry | thank you, I'll do it |
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| 19:49.49 | sebastienthiry | hup, here is my log from asterisk, there is just this two lines in my /var/log/messages https://pastebin.com/WkQpJeqe |
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