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| 02:26.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Security Only: 15.6.1 (2018/09/20); 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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| 10:16.09 | lwlvl | is there a way to place a call by asterisk itself by using the soundcard? (something like an integrated softphone) or do I need to install something like linphone in cli-mode? |
| 10:16.49 | wakko1 | i think you are looking for chan_console |
| 10:17.26 | lwlvl | wakko1, okay...I'll have a look at it :) |
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| 10:18.23 | lwlvl | wakko1, the idea is, to have asterisk call a phone and use it's own soundcard (where door-opener-audio is installed...) |
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| 10:19.03 | wakko1 | chan_console should do |
| 10:21.11 | wakko1 | you also have chan_alsa i think |
| 10:21.35 | wakko1 | they are pretty much the same |
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| 10:23.21 | sibiria | chan_console should be the portable one that works across multiple OSes and audio frameworks |
| 10:23.33 | sibiria | macOS, windows, most *nixes |
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| 10:32.32 | lwlvl | thank you, guys :) |
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| 16:24.55 | Kobaz | soooo |
| 16:25.09 | Kobaz | how would STIR/SHAKEN work with respect to call-forwarding and preserving original callerid |
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| 17:00.54 | *** topic/#asterisk by rmudgett -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 17:06.38 | [TK]D-Fender | rmudgett, 16 = LTS? |
| 17:06.45 | file | yes. |
| 17:06.54 | [TK]D-Fender | yay |
| 17:08.05 | file | we had some network difficulties so we're slow getting everything out/updated |
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| 19:12.08 | igcewieling | heh, uptime 1542 days |
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| 19:47.42 | Samot | On the one hand, impressive. On the other, not so sure. |
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| 20:40.26 | Kobaz | anyone else getting "Updater has an incorrect signature" with the latest polycom vvx firmware 5.8.1 ? |
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| 21:08.45 | xyz111 | Hi Guys, I'm a noob to asterisk, and want to know, how difficult is it to set up a conference call bridge with it? Can I set it up so users can call in with a standard landline number? Can the landline number be used by multiple people at the same time? Thanks :) |
| 21:10.39 | Samot | Landlines are generally limited to two calls |
| 21:13.14 | xyz111 | how does it work when I dial into a call on gotomeeting? Sometimes the same number is being used by 20 people. Do they do something special with their lines? |
| 21:16.27 | Samot | Now a days? SIP. |
| 21:16.37 | Samot | Before that, PRIs. |
| 21:17.02 | xyz111 | are there any good links on how this works? I'd love to learn how to do it |
| 21:17.18 | Samot | freepbx.org/downloads |
| 21:17.38 | Samot | Go download FreePBX. It is a PBX system built on Asterisk. |
| 21:17.47 | xyz111 | thank you |
| 21:17.53 | Samot | You can configure your conference rooms through the GUI. |
| 21:18.03 | xyz111 | will this allow me to receive multiple calls at once? |
| 21:18.05 | Samot | And setup SIP trunks with a provider. |
| 21:18.18 | xyz111 | ah, so I need to get my provider involved |
| 21:18.23 | Samot | Of course. |
| 21:18.40 | Samot | You still need a connection to the PSTN to make and receive calls outside of the PBX. |
| 21:19.13 | xyz111 | is this usually expensive? for example, if I need to have upto 50 people on a call, do I pay 50 "line rentals" to my provider? |
| 21:19.38 | xyz111 | sorry if that's a silly question |
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| 21:31.22 | Samot | Most SIP providers charge by the minute so they don't overly care how many calls for the most part. |
| 21:33.36 | xyz111 | ah interesting... do you think a web based support service using something like webrtc might replace the need for SIP and similar techs? |
| 21:34.00 | Samot | No. |
| 21:34.11 | Samot | WebRTC uses SIP |
| 21:34.23 | Samot | It's just a different method a delivery. |
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| 23:43.47 | PixelAsterisk | Hey guys! I know I can get uniqueid via AGI in my bash script via ${agi_uniqueid}, but can you get linkedid via AGI? I couldn't find it in any of the references. |
| 23:46.39 | [TK]D-Fender | it's a variable like any other... |
| 23:48.12 | [TK]D-Fender | https://community.asterisk.org/t/re-asterisk-13-3-2-how-get-linkedid-solved-and-sequence/48629 |
| 23:49.31 | PixelAsterisk | I'm likely misunderstanding, but I know I can call AGI(nameofmyscript.agi, ${CHANNEL(LINKEDID)}) and pass it that way, but normally with AGI's I can just call the script AGI(nameofscript.agi), and from within that reference ${agi_variable} and get the environment variables that way |
| 23:49.51 | PixelAsterisk | however linkedid doesn't seem to be one of them, whereas uniqueid is. Am I missing something obvious? (highly likely!) |
| 23:49.56 | [TK]D-Fender | no |
| 23:50.12 | [TK]D-Fender | you get it the same way you would get any other regular channel variable |
| 23:51.35 | PixelAsterisk | ok thanks, appreciate your time! |
| 23:51.52 | [TK]D-Fender | You don't have to pass variables. |
| 23:51.56 | [TK]D-Fender | You just grab them direct |
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| 23:55.57 | PixelAsterisk | Thanks for the help D-Fender |