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| 01:41.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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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| 13:14.07 | stevenm | an eerie quiet... |
| 13:15.17 | pawiecki | stevenm: not anymore :) |
| 13:16.26 | stevenm | pawiecki, think sangoma will do good by the * community? |
| 13:17.50 | sibiria | fork while you can!1 |
| 13:19.00 | pawiecki | stevenm: what do you mean - I'm out of the loop |
| 13:19.19 | Samot | Huh? |
| 13:19.32 | stevenm | https://www.sangoma.com/press-releases/sangoma-announces-definitive-agreement-to-acquire-digium-inc/ |
| 13:21.08 | Samot | Oh. |
| 13:21.12 | Samot | Shrug. |
| 13:22.41 | stevenm | I'd love to see FreePBX get named back to Asterisk Management Portal :P |
| 13:22.55 | stevenm | just for the fun of it - no conflict in the naming any more |
| 13:23.42 | Samot | Why? |
| 13:23.55 | Samot | FreePBX is one of many that use Asterisk as their underlying voice engine. |
| 13:24.09 | Samot | Also, why would FreePBX drop their own branding? |
| 13:24.19 | stevenm | you're reading too much in to it |
| 13:24.40 | wyoung | .o/ |
| 13:24.48 | Samot | I'm not reading too much in to it. |
| 13:27.42 | tm1000 | stevenm: yes. We will :) |
| 13:28.03 | tm1000 | We will have an FAQ out shortly. Not quiet. Just planned. |
| 13:28.43 | sibiria | will asterisk 16 be LTS? |
| 13:28.48 | sibiria | once it comes out of RC |
| 13:28.50 | file | Timezones are a fun thing |
| 13:28.54 | file | sibiria: yes, 16 will be LTS |
| 13:28.57 | cresl1n | sibiria: yes. |
| 13:28.58 | cresl1n | oops |
| 13:29.01 | sibiria | cool, thanks |
| 13:29.03 | cresl1n | Thanks file :-) |
| 13:29.09 | sibiria | any rough ETA or so? |
| 13:29.14 | cresl1n | for 16? |
| 13:29.15 | sibiria | yeah |
| 13:29.19 | file | sibiria: you can see the schedule and release dates for things on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:29.20 | cresl1n | Typically Astricon |
| 13:29.31 | stevenm | tm1000, I was just kidding :P |
| 13:29.34 | file | (that includes past so you know when old versions go security only or EOL) |
| 13:29.35 | sibiria | thanks, file |
| 13:30.25 | file | for 16 they are goals and may change depending on the future |
| 13:30.47 | stevenm | tm1000, although I do think if asterisk.org could become a commercial-free zone for both asterisk and freepbx/amportal communities together... that'd be nice. like an "Asterisk Foundation" akin to the Apache Foundation. would prevent alienating alot of people worried that sangoma will just merge everything together and continue down the open-core route |
| 13:31.46 | Samot | But neither Digium or Sangoma are commercial free. |
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| 13:32.00 | stevenm | Samot, I never said they were. |
| 13:32.10 | Samot | The communities do just fine now |
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| 13:32.17 | Samot | No one is going to change that. |
| 13:32.19 | stevenm | you missed your full stop |
| 13:32.22 | stevenm | oh it's back :P |
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| 13:32.45 | Samot | You mean the period at the end of a sentence? Like proper grammar? Yes, I missed it on one sentence. |
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| 13:35.03 | stevenm | Samot, you don't remember any of this I'm guessing? |
| 13:35.47 | Samot | I do. |
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| 13:39.43 | pawiecki | Well, basically periods are used to separate sentences. If you're writing your last sentence, then period is more of a visual (then functional) character. |
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| 14:12.15 | angryuser | So, Sangoma (FreePBX) to Acquire Digium (Asterisk) for $28M |
| 14:12.23 | angryuser | if anyone missed |
| 14:12.54 | file | for anyone who may have questions there is an FAQ at https://wiki.asterisk.org/wiki/display/AST/Sangoma+and+Digium+Join+Together+FAQ |
| 14:13.55 | aoeui | wow |
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| 14:26.51 | pawiecki | file: cool, thanks. |
| 14:27.02 | file | you're welcome! |
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| 14:30.30 | pawiecki | malcolmd: your "Baby Got Back" parody (or should I say cover) made my day! |
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| 14:46.02 | malcolmd | pawiecki :D |
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| 14:57.32 | pawiecki | malcolmd: "Sir Mix-a-Malcolm" haha! :D |
| 15:00.05 | pawiecki | Anyway, have a nice weekend! Bye, all. |
| 15:06.05 | igcewieling | is disappointed by the merger. If Sangoma didn't have good cards competing with Digium, Digium would never have fixed the issues with their cards. |
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| 15:32.50 | seanbright | and why exactly does that disappoint you? |
| 15:33.18 | seanbright | company A has something good, company B has something with issues, company A buys company B. |
| 15:33.43 | seanbright | seems like that would be a good net result |
| 15:34.29 | Samot | I believe it is the logic of "less competition equals less innovation" |
| 15:35.12 | sibiria | my only impression of sangoma is what i got trying to use their Lyra AMD with asterisk |
| 15:35.21 | sibiria | horrible integration, very poor accuracy |
| 15:35.26 | sibiria | terrible product |
| 15:35.39 | sibiria | was worse than LumenVox, and that says a lot |
| 15:40.20 | igcewieling | Digium cards have not been crap for 8 - 10 years, but they would never have improved if someone else like Sangoma wasn't competing. |
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| 15:43.43 | seanbright | got it. so your contention is that the lack of competition will result in lower quality product. |
| 15:45.18 | Samot | It is a valid concern. |
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| 15:46.40 | igcewieling | seanbright: yes |
| 15:48.51 | anbraten | Can I set a username for login which is different from the sip extension number? Because I don't wont to change the client settings each time I give users a different local number. |
| 15:48.54 | Samot | For mean, the issue with cards being good or bad is not even a concern. I don't use them. |
| 15:49.27 | Samot | anbraten: You need to be a little more clearer. |
| 15:50.09 | Samot | The SIP credentials have nothing to do with what Asterisk considers "extensions" |
| 15:50.22 | Samot | I can dial 200 and ring a device with the username bob |
| 15:51.11 | Samot | exten => 200,1,Dial(SIP/bob) |
| 15:51.15 | Samot | exten => 210,1,Dial(SIP/bob) |
| 15:51.25 | Samot | ^^ Both call bob |
| 15:51.55 | seanbright | my only experience with sangoma was with their driver software which sat on top of zaptel/dahdi |
| 15:52.07 | seanbright | and that was painful |
| 15:52.12 | seanbright | granted that was a decade ago |
| 15:52.15 | anbraten | I have exten => _XXX,1,Dial(SIP/${EXTEN}), but want my phone to register with bob |
| 15:52.17 | seanbright | so surely times have changed |
| 15:52.57 | igcewieling | anbraten: don't do that. |
| 15:53.32 | Samot | Then you can't do Dial(SIP/${EXTEN}) |
| 15:53.44 | anbraten | <igcewieling>: whats the problem? |
| 15:53.57 | igcewieling | anbraten: you are using the extension as the username. |
| 15:54.17 | igcewieling | exten => _XXX,1,Dial(SIP/bob) if you want to dial bob |
| 15:55.17 | Samot | "bob" would be the section in sip.conf called [bob] |
| 15:55.27 | Samot | Which could have username=bob or username=1000 |
| 15:56.27 | anbraten | Mhh, but that way I need to configure each sip extension myself. |
| 15:57.30 | igcewieling | yes, that is correct. If you don't want to do that then keep the extension and username the same. |
| 15:59.06 | igcewieling | you could also do like freepbx does, use a database, AGI scripts, and custom dialplan, it works well enough but is a lot of work. |
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| 16:18.54 | babak_ | Hi, I want to dial simultaneously 2 channels but bridge just channel one early media for caller, is it possible ? |
| 16:19.29 | jsmith | babak_: The only way I can think of doing that is with a Local channel |
| 16:20.23 | jsmith | (on just one of the two endpoints being dialed) |
| 16:23.02 | anbraten | <igcewieling>: okay. thanks |
| 16:23.14 | anbraten | But what is the difference of username=bob and the [bob] head? |
| 16:24.54 | babak_ | jsmith: thx for bridging early media there is an option in dial application ? |
| 16:29.20 | Samot | babak_: You can't. |
| 16:30.49 | Samot | babak_: You would have to Dial(SIP/endpoint1&SIP/endpoint2) to dial them simultaneously. |
| 16:31.21 | Samot | The option R for playing ringback but allowing for early media to interrupt is a default Dial() option. |
| 16:31.51 | Samot | It happens on all endpoints being Dial()'d. |
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| 17:03.44 | babak_ | Samot: thx, if with local channels we dial multiple channels , we still can not hear early media of dialed channels(or onr of them) ? |
| 17:03.51 | babak_ | onr/one |
| 17:05.03 | Samot | Are you doing that now? |
| 17:05.42 | babak_ | no tomorrow I can test |
| 17:05.55 | Samot | Dial(), by default sends ringback |
| 17:06.20 | Samot | And if early media is detected, it will use the early media |
| 17:06.35 | Samot | But its on *all* channels |
| 17:07.05 | Samot | You cant dial two at once and then pick and choose which one responds to early media |
| 17:07.12 | babak_ | for parallel calls it seems there is no bridge early media or mix early media |
| 17:07.16 | Samot | They both will |
| 17:07.32 | Samot | Early media isnt an answered call |
| 17:08.15 | babak_ | https://issues.asterisk.org/jira/browse/ASTERISK-22201 |
| 17:08.42 | babak_ | mentioned: in a parallel call, Dial() will never bridge any early media audio, |
| 17:10.31 | Samot | Then use the r option |
| 17:10.43 | Samot | Thats for ringback |
| 17:10.54 | Samot | The ticket |
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