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| 02:36.23 | igcewieling | mutters something about zombie werewolves, |
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| 02:56.16 | tcpdump | Hello everyone. |
| 02:56.41 | tcpdump | Anyone know a good diagram or doc that explains the architecture of a basic PBX/asterisk setup |
| 02:57.02 | voipmonk | Basic ? - explain |
| 02:57.05 | tcpdump | Specifically all of the components such as a TURN server, asterisk server, etc. |
| 02:57.26 | voipmonk | google images asterisk pbx |
| 02:57.55 | voipmonk | not all implementations require a TURN server |
| 02:58.19 | lorsungcu | tcpdump: do you have a specific concern? |
| 02:58.53 | tcpdump | lorsungcu: not really, just trying to figure out the pieces, what they do, etc. |
| 02:59.20 | lorsungcu | that's entirely up to you |
| 02:59.42 | voipmonk | what are you looking to do with asterisk ? |
| 02:59.46 | lorsungcu | figure out what it is you want to do and start doing it. usually you'll figure out what's needed as you go. |
| 02:59.51 | tcpdump | Im just trying to wrap my head around the concept at a high level, then drill into the different pieces. I was hoping get some guidance into what a basic setup looks like. |
| 03:00.31 | lorsungcu | you dont need more than a server running asterisk |
| 03:00.40 | tcpdump | I know what I want. I want a remote server that facilitates audio/video conference calls. All clients will be remote to the server. |
| 03:01.23 | Samot | Well |
| 03:01.33 | voipmonk | how many conferences and how many people per type of conference? |
| 03:01.41 | Samot | Asterisk doesnt support video confbridges |
| 03:01.51 | Samot | So there is that |
| 03:02.09 | tcpdump | So it will be on the WAN, in Amazon for instance, and all clients will be able to connect and call each other from anywhere, so long as they have Internet. |
| 03:02.38 | Samot | If video conf is a deal breaking requirement, Asterisk is not the solution. |
| 03:02.41 | tcpdump | voipmonk: at any given time there will never be more than one to one on a call, but there may be many calls. |
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| 03:03.00 | tcpdump | I see Samot . |
| 03:03.15 | tcpdump | Are you aware of any other ip conference solutions that do video? |
| 03:03.25 | Samot | Even then, i wouldnt recommend AWS |
| 03:03.35 | Samot | there are a few |
| 03:03.36 | lorsungcu | tcpdump: skype works p. good |
| 03:03.47 | Samot | 3CX |
| 03:03.58 | tcpdump | Samot: yea, that was an exmaple. Likely wont be aws. |
| 03:04.09 | tcpdump | lorsungcu: yea, needs to be a private solution though. |
| 03:04.15 | tcpdump | 3cx huh? |
| 03:04.17 | tcpdump | lemme google that. |
| 03:04.31 | Samot | Its licensed. |
| 03:04.44 | Samot | So you are going to spend $$$ |
| 03:05.03 | tcpdump | Ah, is that basically a paid version of freepbx? |
| 03:05.06 | Samot | You need voice license and video conf license. |
| 03:05.11 | Samot | No |
| 03:05.15 | Samot | Not at all |
| 03:05.16 | voipmonk | have a look at https://jitsi.org/downloads/ you can use existing asterisk trunks - FREE |
| 03:05.29 | Samot | .... |
| 03:05.31 | Samot | What |
| 03:06.01 | lorsungcu | its a masonic tax evading thing |
| 03:06.04 | lorsungcu | you wouldnt understand |
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| 03:07.26 | tcpdump | Anyone used 3CX before? |
| 03:07.31 | tcpdump | They pretty good? |
| 03:08.05 | Samot | Like i said...$$$$ |
| 03:09.11 | Samot | You need to research |
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| 03:14.16 | Samot | 3CX might be out of price range but Asterisk does not have the video conf support you require. |
| 03:15.14 | lorsungcu | the "must be private" but "hosted on AWS" requirement is a little bizarre |
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| 06:47.43 | snadge | just playing around with asterisk 14, as you do.. looks like the split cdr bug with respect to the hangup handler has been fixed |
| 06:47.59 | snadge | which is great |
| 06:48.13 | snadge | now i just need to figure out why CDR(dst) is being set to "s" instead of the destination number ;) |
| 06:57.34 | snadge | because a macro is overwriting it.. if i bypass the macro, it works as expected |
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| 07:03.48 | snadge | that can be put off until tomorrow ;) |
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| 08:49.02 | _8bits | Is there a way to use only one GoToIfTime if you want to use it from jul 1 to aug 14 as example? |
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| 09:06.13 | jbrouwers | Hello, is ARI always supposed to send out an event when a call recording stops? ie either RecordingFinished or RecordingFailed? |
| 09:07.01 | jbrouwers | I am looking at my logs, and I got 2439 RecordingStarted events, but only 378 RecordingFinished events. |
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| 09:21.54 | dinesh85 | Hi, can someone tell me if it is possible to configure asterisk for priority based call handling? Example: When a same extension is dialed by two users(User A & User B), that User A has more priority than User B. This mean if there is an existing call from User B to an extension, and if User A dials the same extension, the call from User B has to be terminated, and continue with the call from User A. |
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| 09:54.25 | dnit | Hi When I bridge my agent inbound channel to another channel , my agent logs out. |
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| 13:16.30 | [TK]D-Fender | dinesh85, you'll have to do some external scripting to look at the calls to determine If they are bridged with someone of a lower class |
| 13:17.24 | [TK]D-Fender | dinesh85, Likely using AMI to look at whatever channel is currently bridged with your target and looking at its channel variables for a value you'd set BEFORE it dialed them which would be set to indicate their priority |
| 13:18.02 | [TK]D-Fender | then via AMI you'd disconnect them if appropriate and you could resume nrmal dialplan to set the priority for this channel and then be free to dial on in. |
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| 17:12.34 | atotclic | hello, buenas tardes |
| 17:14.05 | atotclic | help!! I can not configure snmp |
| 17:14.49 | Samot | Uhm. |
| 17:14.57 | Samot | Wrong channel? |
| 17:16.14 | atotclic | Samot: this cannel is asterisk |
| 17:16.28 | Samot | Yes. |
| 17:16.32 | atotclic | problem with module res_snmp.so |
| 17:16.40 | Samot | OK, that's different. |
| 17:16.55 | atotclic | module is loaded |
| 17:17.30 | atotclic | snmpwalk not response |
| 17:18.39 | Samot | https://wiki.asterisk.org/wiki/display/AST/Simple+Network+Management+Protocol+%28SNMP%29+Support |
| 17:18.40 | atotclic | Samot: sorry my english |
| 17:18.49 | Samot | ^^ That's the best I can offer. I don't use SNMP with Asterisk. |
| 17:21.38 | atotclic | Samot: thank you, but it does not work |
| 17:26.06 | atotclic | asterisk version 14.6.0 |
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| 18:15.58 | atotclic | Cannot adopt OID in STERISK-MIB: astChanState ::= { astChanEntry 23 } |
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| 18:30.26 | tcpdump | anyone know of a guide to help understand the exten => values in the config? |
| 18:31.33 | Samot | ~book |
| 18:31.34 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:32.37 | Samot | I take it video conference is no longer a requirement? |
| 18:41.55 | shanth | tcpdump: where are you stuck? |
| 18:42.04 | shanth | tcpdump: im probably a few days ahead of you |
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| 18:47.11 | tcpdump | shanth: thanks - im just trying to find a page or man that explains the structure and function of these config values: exten => 6001,1,Goto(6002,1) |
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| 18:48.13 | shanth | ah yeah best bet is to start reading one of the asterisk book resources tcpdump |
| 18:48.31 | shanth | lots of them go over in detail how the dialplan works, it's kind of the heart of asterisk |
| 18:48.39 | tcpdump | ah |
| 18:48.45 | tcpdump | its called a dial plan |
| 18:48.46 | tcpdump | got it |
| 18:48.56 | tcpdump | now i know what to google. :) |
| 18:49.48 | [TK]D-Fender | No need to google |
| 18:49.56 | [TK]D-Fender | there's a book. You've been linked to it already |
| 18:55.15 | Samot | So again, video conferencing is not a requirement anymore? |
| 18:55.37 | Samot | Because yesterday this was about a audio/video conferencing solution. |
| 18:56.58 | Samot | tcpdump: Have your requirements changed? Because if not, my statement from yesterday still stands. Asterisk will not do video conferencing at this time. |
| 18:57.30 | tcpdump | Samot: no - however, video seems to be working just fine. |
| 18:57.38 | Samot | In conference? |
| 18:57.41 | tcpdump | I can't tell you why - but it is. |
| 18:57.42 | tcpdump | Yes |
| 18:57.47 | Samot | There is a difference between a video call and a conference call? |
| 18:57.55 | Samot | How are you conferencing these video calls? |
| 18:58.02 | tcpdump | I need video calls |
| 18:58.05 | tcpdump | not conference. |
| 18:58.16 | Samot | That's not what you said yesterday. |
| 18:58.19 | tcpdump | I suppose maybe I never had that right then? |
| 18:58.35 | Samot | Asterisk will support video calls.. |
| 18:58.44 | Samot | But not video conference bridges. |
| 19:01.21 | Samot | file: Now I'm curious. Can Asterisk support "3-Way" or "N-Way" video calls? |
| 19:01.50 | shanth | also tcpdump if you are in such a mood there exists a cheap udemy course for $10 that covers a lot of dialplan basics in video format. i found it helpful |
| 19:02.00 | file | ConfBridge currently supports 1 channel being the source of video, it is then sent to the other participants |
| 19:02.10 | Samot | OK |
| 19:02.13 | shanth | since i am now administrating an asterisk system as of monday i needed to get up to speedquickly |
| 19:02.24 | Samot | But what about 3-way video calls? |
| 19:02.42 | Samot | No bridge, just at the phone/client doing it. Like an audio call... |
| 19:03.10 | file | that's up to the client. |
| 19:03.23 | file | to Asterisk those are separate calls, the mixing and display is client side |
| 19:03.23 | Samot | OK. |
| 19:03.38 | Samot | That's why I thought. |
| 19:03.46 | Samot | s/why/what/ |
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| 19:08.42 | Samot | tcpdump: So if all you need is video calls or at least one video call in the bridge that everyone else can stream. Asterisk is fine. |
| 19:09.12 | Samot | tcpdump: I just want to make sure you don't spend time banging away at Asterisk to realize it really doesn't do what you need. |
| 19:09.45 | tcpdump | Samot: I apologize man, for being so ignorant. This is a whole new world to me. |
| 19:10.00 | Samot | No need for that. |
| 19:10.11 | tcpdump | Yea, I only ever need a single video call, not a conference. |
| 19:10.17 | Samot | Like I said, I just want to make sure you understand what it does compared to what you require. |
| 19:10.25 | Samot | Clarifying the requirements isn't a bad thing. |
| 19:10.27 | tcpdump | I do recall asking for the other, not realizing the difference. |
| 19:10.49 | Samot | Then yes, Asterisk supports video calls. |
| 19:10.55 | Samot | So it can do what you are looking for. |
| 19:11.28 | tcpdump | Yea, I have it up and mostly working - I just have a few other little quirks to iron out, and I think thy're starting to make sense. :) |
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| 19:55.03 | tcpdump | So it looks like when a user registers it sends a REGISTER, that gets a 401, then it sends a second register with the nonce from the 401 response and gets registered the second time? |
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| 20:11.09 | dadrc | that's what's supposed to happen, yeah |
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| 22:29.50 | darkunderlord | i have no freaking idea, but trying to set my 13 server live and when dialing certain numbers I'm getting the demo-congrats. I've looked/searched my extensions.conf and demo isn't even in there.... |
| 22:33.11 | rmudgett | What about the default extensions.ael or extensions.lua? Have you removed them if you aren't using them? |
| 22:35.34 | darkunderlord | hmm maybe accidentiall |
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