| 17:12.00 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:12.00 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.26.0 (2019/4/4) 16.3.0 (2019/4/4), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 17:20.25 | Samot | Nope. |
| 17:21.57 | Samot | wonderworld: In regards to SIP, echo cancellation isn't a thing. |
| 17:22.16 | Samot | FXS/FXO gateways may have that option but that is for the FXS/FXO side of it. |
| 17:22.21 | Samot | Not the SIP side. |
| 17:23.21 | wonderworld | ok. i haven't tried it out yet, but i fear the person on the sip peer would hear herself, because her voice would bounce back to the mic on alsa side. |
| 17:25.44 | sibiria | vn971: to be fair, the default config of asterisk is |
| 17:25.45 | Samot | So you have person A on chan_alsa and person B on chan_sip, what is the wrory again? |
| 17:25.50 | sibiria | *almost* ready to go for what you're asking |
| 17:26.04 | sibiria | but if you're looking for something even easier, check out freepbx |
| 17:26.06 | Samot | That person A is going to hear themselves? |
| 17:26.10 | Samot | No. |
| 17:26.14 | Samot | Chan_alsa isn't supported. |
| 17:26.32 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 17:26.38 | Samot | FreePBX doesn't support all the random and low use stuff like gsm, alsa, etc. |
| 17:27.00 | Samot | You'd be in the same boat of doing it all manually. |
| 17:27.10 | Samot | On top of hooking it into pre-generated stuff. |
| 17:27.11 | sibiria | Samot: no my message was for vn971 and their use-case of just hooking phones to asterisk to call eachtoehr |
| 17:27.37 | Samot | 1:04:22 PM <Samot> #freepbx |
| 17:27.37 | Samot | 1:04:51 PM <Samot> Full GUI system that installs and manages Asterisk for you. |
| 17:27.37 | Samot | 1:05:08 PM <Samot> Point, click, fill out forms, submit, apply, done. |
| 17:27.39 | Samot | Ahh that |
| 17:27.46 | wonderworld | Samot: i use asterisk |
| 17:27.59 | sibiria | right, you already adviced them of freebx |
| 17:28.28 | Samot | wonderworld: I know. So what is your worry again? |
| 17:28.37 | Samot | Who is going to hear themselves? Person A or Person B? |
| 17:28.44 | vn971 | sibiria: how do I even install it though? `apt search freepbx` gives zero hits, search engine gives articles starting with > Manual installations of FreePBX is considered an EXPERTS ONLY exercise. |
| 17:29.06 | vn971 | * feels like a newbie, even though has been on Linux ~ 10 years. |
| 17:29.25 | Samot | freepbx.org/download |
| 17:29.49 | sibiria | vn971: it's a self-hosted product. it's built on top of CentOS |
| 17:29.58 | Samot | You have two options, use their ISO which does everything for you or manually install it over Asterisk but then you have to do a bunch more stuff. |
| 17:29.58 | wonderworld | person A is on the SIP leg. person B on alsa. on the alsaside is no headset. there is a standalone mic and a standalone speaker. i fear that perosn A would hear herself when talking, beause her voice would go back thru the mic on the alsa side. |
| 17:30.13 | vn971 | Samot: but that's so... unusual? I feel like somebody asks me to download an exe file. |
| 17:30.25 | vn971 | or maybe it even is in fact a statically compiled ELF ? |
| 17:30.29 | Samot | A project having an ISO? |
| 17:30.36 | Samot | That's unusual? |
| 17:30.39 | Samot | No, it's not. |
| 17:30.52 | Samot | FreePBX has it's own version of RHEL. |
| 17:30.58 | Samot | It runs it's own repos. |
| 17:31.18 | Samot | It doesn't use standard repos any RHEL would use. |
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| 17:31.46 | vn971 | Samot: and the fact that I'm on Debian doesn't matter for it I guess, right? |
| 17:31.54 | Samot | It won't matter. |
| 17:32.03 | sibiria | vn971: as said, freepbx is a self-hosted product. if you need to achieve your PBX on some sort of shared setup where you run asterisk yourself, it will be easier to just configure asterisk for it, and you will need to do a little bit more than just setting up the various user peers |
| 17:32.09 | Samot | 1:30:53 PM <Samot> FreePBX has it's own version of RHEL. |
| 17:32.30 | Samot | If you don't use the ISO you can manually install it on Debian, Ubuntu, RHEL, CentOS |
| 17:32.40 | Samot | But then YOU HAVE TO DO ALL THE WORK |
| 17:33.00 | Samot | You have to make sure the server has everything it needs. All the proper dependencies, et all. |
| 17:33.33 | Samot | wonderworld: Does her voice come out the speak when she uses the mic normally? |
| 17:35.03 | wonderworld | i haven't built it yet, so no idea. i am planing and was wondering if an echo problem would be solveable at all. |
| 17:35.34 | vn971 | sibiria: is there some web page that describes what else needs to be touched to have it all configured? For my type of use case. |
| 17:35.35 | Samot | Well first it's not going to be an echo problem. |
| 17:35.51 | Samot | You're worried about speakers kicking out her voice over herself while she talks. |
| 17:35.55 | vn971 | Also wouldn't want to steal your time folks of course if I should instead be looking into some other solution. |
| 17:35.59 | Samot | And the other side hearing that. |
| 17:36.04 | Samot | Because they would. |
| 17:36.12 | Samot | To them it sounds like an echo but it's not. |
| 17:36.24 | Samot | You'd be piping sound through the speakers. |
| 17:36.30 | sibiria | vn971: uh surely, but i'm not very familiar with any. it's ok to ask here, though. it's the right place. you should begin with setting up users for pjsip and just test dialing in to a pre-made dial plan playing back some sound or so, so you know that part's working before you start connecting users |
| 17:36.33 | Samot | Get them a proper headset if you're going to do this. |
| 17:36.47 | sibiria | vn971: so i can give you a simple example of how to achieve that to get you started |
| 17:37.17 | Samot | sibiria: Of everything? |
| 17:37.39 | vn971 | sibiria: let me try to process some of it first.. Big thanks for all the answers though! |
| 17:38.33 | sibiria | Samot: to be fair, without approacing this as if vn971 was building a production grade product it's *mostly* a case of setting up users in pjsip |
| 17:38.40 | Samot | wonderworld: Or better yet, don't have them directly on the PBX. |
| 17:39.00 | Samot | sibiria: And modules.conf, rtp.conf, CEL, CDR, etc, etc, etc. |
| 17:39.13 | Samot | There's more to this than just extensions.conf and pjsip.conf |
| 17:39.29 | sibiria | he's setting something up to just let a few buddies call eachother |
| 17:39.33 | sibiria | not building a PBX product |
| 17:39.35 | Samot | I know what it is. |
| 17:39.39 | Samot | But there are still basics. |
| 17:43.07 | sibiria | not sure if CEL + CDR will be needed...unless they plan on billing eachother :) |
| 17:43.20 | Samot | Or what to look at historys. |
| 17:43.22 | Samot | Troubleshoot. |
| 17:43.31 | Samot | But sure. |
| 17:43.58 | Samot | Point I'm trying to prove: You linked him to the PJSIP wizard stuff... |
| 17:44.14 | Samot | He was confused by the template parts of the config. |
| 17:44.32 | Samot | Because Templates are a different thing that apply globally and not to just PJSIP. |
| 17:44.43 | sibiria | for the simple case of a teamspeak facsimile for a few friends to use, the bulk of the effort needed is to set users up in pjsip |
| 17:44.49 | Samot | So we're back to the basics. |
| 17:45.00 | Samot | You're missing my point. |
| 17:45.07 | Samot | I'm encouraging the user to learn the basics. |
| 17:45.17 | Samot | Not give them something they can copy and paste and be ready to go. |
| 17:45.22 | Samot | Because nothing is learned. |
| 17:46.58 | sibiria | i'll refrain from assuming anything about their methods of learning |
| 17:48.42 | Samot | I don't care what methods they use. |
| 17:48.47 | Samot | As long as they are doing it. |
| 17:49.16 | Samot | If hopping on one leg while humming the theme to Golden Girls helps you absorb knowledge. Hop and hum away. |
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| 18:03.48 | tnt | Hi everyone. Anyone familiar with dahdi / wcte1xxp / te110p ? |
| 18:04.19 | tnt | I'm trying to prevent it from locking to the far end bit-clock. (i.e. I want it to be the timing master of the link). |
| 18:05.54 | tnt | I tried timing_source=0 and 1 in system.conf but in both case it seems to lock onto the incoming bit clock. |
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| 18:07.33 | Samot | Not I. I don't (or let friends) use DAHDi. |
| 18:07.59 | tnt | What do you use ? |
| 18:08.06 | Samot | SIP. |
| 18:08.32 | tnt | Ah well that's not going to work for me :) |
| 18:09.11 | Samot | There are a few DAHDi users that are active in here but it is Easter Sunday so no guarantees on replies. |
| 18:14.17 | tnt | yeah, I 'm not even sure what I want to do is possible at all. |
| 18:22.11 | drmessano | tnt: Did you try setting 0 for timing in the span definition? |
| 18:23.17 | drmessano | What does your span line look like? |
| 18:24.29 | tnt | I tried "span=1,1,0,ccs,hdb3,crc4" and "span=1,0,0,ccs,hdb3,crc4" running `dahdi_cfg -c /etc/dahdi/system.conf` after each change. |
| 18:25.42 | tnt | drmessano: ^^ |
| 18:27.12 | drmessano | The second would be my choice.. Not sure i've ever reloaded that way before.. I usually restart the service |
| 18:32.10 | tnt | if I change the bchan/dchan config, I can see it changed in /proc/dahdi/1 after running dadhi_cfg so I think it's applied. |
| 18:34.40 | wonderworld | tnt: be very careful. i wasted hours of time because changes i were trying out never were applied without reloading the modul, but that was years ago. |
| 18:40.10 | tnt | wonderworld: I think you're right ... just unloaded the module and reloaded it and reddid dahdi_cfg and now the transmit bit clock of the card doesn't lock to the incoming bit clock. |
| 18:44.09 | tnt | yup confirmed it with the scope. |
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| 21:49.16 | drmessano | wonderworld: yep been there too |
| 21:49.31 | drmessano | I know I at least restarted the service. Maybe more. |
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| 23:02.49 | life_of_e_ | How to scare everyone near a phone: 1. Connect an NWS weather radio audio out to a SIP intercom adapter, 2. connect the alert LED to the intercom dial button, 3. set up an extension to auto page all phones, 4. don't tell anyone about it, 5. wait for the weekly test. |
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