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| 02:37.09 | tzafrir | Can anybody think of prior art to https://www.google.com/patents/US8914003 ? |
| 02:37.31 | tzafrir | See https://www.eff.org/deeplinks/2016/04/stupid-patent-month-voice2text-trolls-voip-providers |
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| 16:42.43 | xpheres | <PROTECTED> |
| 16:42.50 | xpheres | I wonder if anyone has tested it and can guide me to apply the patch |
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| 17:20.43 | btcNeverSleeps | hi everybody, anyone here on a sunday? (trying to add a SIP Phone to a Raspberry / RasPBX install which already has working phones) |
| 17:20.54 | btcNeverSleeps | is working on a sunday :-/ |
| 17:27.35 | *** join/#asterisk herrkin (~herrkin@200.8.137.14) |
| 17:28.36 | herrkin | hello awesome community |
| 17:28.56 | herrkin | I have a problem with my test asterisk server |
| 17:29.06 | herrkin | I stopped listeing to anything on that server |
| 17:29.47 | herrkin | the calls are answered, asterisk shows to play hello-world, or other audio files but I hear nothing on the softphones |
| 17:30.05 | herrkin | I thought it was a problem on the phone or the system (ubuntu) but I have tried several machines |
| 17:30.07 | herrkin | the same thing. |
| 17:30.29 | herrkin | it seems to be an error on the asterisk server that I dont know about. |
| 17:30.44 | herrkin | could you help me? |
| 17:31.03 | btcNeverSleeps | herrkin: sadly I cannot help you, I'm beginning myself |
| 17:35.28 | [TK]D-Fender | "error on the server" really says nothing |
| 17:35.32 | [TK]D-Fender | No audio is a NETWORKING issue |
| 17:36.01 | [TK]D-Fender | either firewalling, or the IP's being offered are not contactable (NAT is the most common thing people misconfigure for) |
| 17:38.12 | btcNeverSleeps | what's the 1000 feet overview as to what needs to be done when you add a 3rd SIP phone to a FreePBX / Asterisk system that already has two working phones? |
| 17:47.38 | [TK]D-Fender | There is none. |
| 17:47.47 | [TK]D-Fender | If you already have 2 working... what's 1000 more? |
| 17:47.49 | [TK]D-Fender | they're all the same |
| 17:47.55 | [TK]D-Fender | There is nothing magical about this |
| 17:48.16 | [TK]D-Fender | Next FreePBX is not supported here, please refer to their channel for support on configuration |
| 17:48.19 | [TK]D-Fender | #freepbx |
| 17:48.27 | btcNeverSleeps | oh gotcha |
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| 20:33.58 | lukeescude | Does anyone happen to know how we could add full transcoding for Opus into Ast 13? |
| 20:35.10 | WIMPy | You could try to push it on the agenda for 14. |
| 20:36.40 | lukeescude | Who would I talk to, to do that? Also when is the development for 14 kicking in? |
| 20:37.13 | WIMPy | Let's try |
| 20:37.17 | WIMPy | ~asterisk-versions |
| 20:37.18 | infobot | Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 20:38.07 | lukeescude | Ha! Love it. Thanks for the info! |
| 20:38.10 | file | the comment by Matt Jordan on http://lists.digium.com/pipermail/asterisk-dev/2013-May/060419.html remains true as of this time |
| 20:38.10 | WIMPy | And IRC is not a bad place, but you should try duting Digium office hours. |
| 20:38.17 | WIMPy | Or the mailing lists. |
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| 20:39.39 | WIMPy | That link might be usefull in the bot. |
| 20:40.57 | herrkin | [TK]D-Fender, you said no audio is networking issue, it was working normally, then it just stopped working. I really havent done much on the server. but on the client I upgraded ubuntu. but as I mentioned I tried with windows or android and still the same. how could I find out what is causing the trouble? |
| 20:41.18 | lukeescude | No audio, or one-way audio? |
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| 21:05.10 | herrkin | I have a problem with my test asterisk server |
| 21:05.12 | herrkin | <PROTECTED> |
| 21:05.12 | herrkin | <PROTECTED> |
| 21:05.12 | herrkin | <PROTECTED> |
| 21:05.12 | herrkin | <PROTECTED> |
| 21:05.13 | herrkin | <PROTECTED> |
| 21:06.44 | herrkin | the asterisk server is a vps I am using softphones on my local network |
| 21:06.51 | herrkin | in case that info helps |
| 21:35.24 | ChannelZ | Some sort of NAT or other routing problem |
| 21:35.42 | ChannelZ | Turn on RTP debug (rtp set debug on) and see where all those voice packets are flinging off to |
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| 22:56.57 | iulhk | hello does asterisk support per minute billing ? if call is connected, can i billed after every minute ? |
| 22:58.11 | Samot | iulhk: Per minute billing is something you decide. There are CDRs generated for the calls, how you use them is up to you. |
| 23:01.23 | iulhk | <Samot>: cdr normally generates, once the call finished, normal asterisk agi, i check user available balance and allow max call duration, and set timeout, so calls finished i check duration and billed user. Now i want during the call after every minute i wanted to deduct cost from user balance, is it possible? during the call can i upgrade timeout value ? |
| 23:02.00 | Samot | iulhk: There are some services for that already or you will have to write something to do what you want. |
| 23:02.26 | Samot | I'm not sure about sending options to a channel once it's established. |
| 23:09.29 | iulhk | <Samot> according to my knowledge, once connection established, you can't send upgrade options to channel :( |
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| 23:25.19 | WIMPy | What are "upgrade options"? |
| 23:25.32 | WIMPy | You can reset TIMEOUT at any time. |
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| 23:25.58 | WIMPy | But IIRC it will count from the time you set it. |
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| 23:33.21 | iulhk | <WIMPy>: even if channel already estabalished, can i reset after every minute? or if i have 200 established connection and i am restting timeout after every minute , does this practice acceptable for asterisk ? |
| 23:42.32 | WIMPy | It's possible. |
| 23:43.34 | WIMPy | It may be acceptable for you. Doing things realtime surely makes things more complicated. |
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