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| 00:16.39 | *** 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: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 02:27.59 | LiuYan | I'm facing malformed character issue when using cdr_adaptive_odbc to write cdr into MySQL database. All UTF-8 encoded Chinese characters are treated as raw bytes and re-encoded again using UTF-8. |
| 02:30.11 | LiuYan | I tired using `isql` to test the datasource configured in odbc.ini, all Chinese characters read/writen are all ok (no malformed character issue) |
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| 08:14.25 | LiuYan | When dial via DAHDI channel, if the called party is ringing, asterisk is not sending '180 Ringing' message to the caller. How to let asterisk send '180 Ringing' message when called party is actually ringning? (I'm not looking for the 'r' option of Dial application) |
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| 10:29.07 | wraythe | Hi everyone :-) |
| 10:29.23 | wraythe | first time hopping into the IRC channel for Asterisk :-) |
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| 10:30.48 | wraythe | I have a weird issue on 16.5.0 |
| 10:31.33 | wraythe | on a call coming in, i can see the INVITE hitting asterisk... Asterisk responds with a 100 Trying, however never follows with a 180 Ringing |
| 10:38.10 | wraythe | Seems like it doesn't actually hit extensions.conf either |
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| 11:23.15 | twanny796 | [2019-07-26 13:20:04] WARNING[612] format_cap.c: Cannot allow unknown format 'g711' |
| 11:24.37 | twanny796 | My ITSP and phone use the g711 codec. Does asterisk use g711? |
| 11:25.19 | drc | There's 2 versions of G.711 |
| 11:25.30 | drc | ulaw and alaw, asterisk can handle both |
| 11:26.46 | twanny796 | The phone only mentions G.711 and the ITSP mentions G.711 and a-Law. |
| 11:27.36 | twanny796 | and I'm getting the error ^^^ in the first line, so I cannot mention g711 |
| 11:27.40 | twanny796 | to asterisk |
| 11:31.26 | twanny796 | [2019-07-26 13:29:01] ERROR[612] res_pjsip_endpoint_identifier_ip.c: Identify 'go-identify' is not configured to match anything. |
| 11:31.26 | twanny796 | [2019-07-26 13:29:01] ERROR[612] res_sorcery_config.c: Could not create an object of type 'identify' with id 'go-identify' from configuration file 'pjsip.conf' |
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| 11:33.23 | twanny796 | The phone uses the g711, the itsp uses 'G.711 a-Law'. Cannot understand why I cannot give g711 to asterisk! |
| 11:35.52 | file | because g711 is not a codec, ulaw and alaw are, so you can enable both of them instead if you want |
| 11:36.27 | file | and the above error states what is the problem with the identify - it's not configured to match anything, so either it's improperly configured or the identify section shouldn't be present |
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| 11:47.14 | twanny796 | file: ok, I had a missing match in the identify, the error is not there now. But when I call the phone rings once and the call is dropped. |
| 11:47.18 | twanny796 | [2019-07-26 13:44:38] WARNING[31901][C-00000002] channel.c: No path to translate from PJSIP/1107-00000003 to PJSIP/go-endpoint-00000002 |
| 11:47.18 | twanny796 | [2019-07-26 13:44:38] VERBOSE[31901][C-00000002] pbx.c: Spawn extension (Go-Incoming, s, 1) exited non-zero on 'PJSIP/go-endpoint-00000002' |
| 11:47.39 | file | the negotiated codecs couldn't be transcoded, so the call was dropped |
| 11:48.04 | twanny796 | problem from the phone? |
| 11:48.18 | file | without knowing the configuration for each side, can't say |
| 11:48.42 | file | and if you don't have codec_ulaw.so and codec_alaw.so loaded then ulaw and alaw can't be transcoded either |
| 11:49.19 | twanny796 | how can I check that? |
| 11:49.49 | file | you look at the configured endpoints.... in pjsip.conf to see what codecs are allowed |
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| 11:50.12 | file | and you can check the modules using "module show like codec" on the command line and seeing if they are loaded. |
| 11:53.34 | twanny796 | I checked with 'core show codecs', and alaw is there. |
| 11:53.48 | file | that doesn't show transcoders |
| 11:54.16 | twanny796 | ok, now alaw is not there |
| 11:54.16 | file | you could also use "core show translation", but first you should provide or check what codecs are actually allowed |
| 11:54.22 | drc | if both your provider and your phone support alaw (which they should), you shouldn't have to transcode |
| 11:54.51 | file | allowing both could yield a scenario where one negotiated ulaw and the other alaw |
| 11:55.26 | drc | sure, possible. but the solution for that scenario isn't transcoding |
| 11:55.47 | drc | file: sorry, that came off as antagonistic. |
| 11:55.59 | drc | Was trying to support your point of checking the configured codecs |
| 11:57.22 | twanny796 | I did 'module load codec codec_alaw.so', and I'm getting incoming calls :) |
| 12:01.21 | twanny796 | file: I put it in modules.conf. Thank you. |
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| 13:25.45 | mtcdood_ | is there a way to simulate an Asterisk setup without having the physical hardware? I'd like to do that. |
| 13:26.42 | Samot | Uhm. VMs work fine. |
| 13:26.49 | Samot | Not sure what else you can do if you don't have that. |
| 13:27.25 | mtcdood_ | yeah you can't actually do anything in a VM though |
| 13:28.19 | mtcdood_ | it looks like EOL'd interfaces are out there for under $200 so it's not a huge investment but money is money |
| 13:28.25 | mtcdood_ | I figured I'd ask first |
| 13:28.51 | Samot | What EOL interfaces? |
| 13:28.54 | Samot | What are you referring to? |
| 13:29.49 | mtcdood_ | an old Digium 4 port analog card with 2 included FXO and 2 included FXS modules |
| 13:30.45 | mtcdood_ | I'd have to ask around and see if they're compatible with Asterisk today but the listing claims they were compatible with some version |
| 13:31.37 | mtcdood_ | but I'm not going to buy stuff any time soon: https://www.amazon.com/Aex410p-Trixbox-aex410-Express-Asterisk/dp/B00H6WJWHM |
| 13:35.09 | Samot | Well you can't simulate FXO/FXS stuff. |
| 13:35.12 | Samot | You need the physical cards. |
| 13:36.27 | [TK]D-Fender | What do you actually want to simulate? |
| 13:36.59 | mtcdood_ | primarily a phone to do some basic stuff like menus |
| 13:37.04 | [TK]D-Fender | What are you trying to validate that would require a card? |
| 13:37.14 | [TK]D-Fender | You don't need PSTN hardware for that |
| 13:37.50 | mtcdood_ | can you give Asterisk DTMF tones as though a physical phone receiver is doing it? |
| 13:37.55 | [TK]D-Fender | You can use a dumb soft-phone on your cell for that and not even need a separate PC from your server. |
| 13:38.18 | [TK]D-Fender | Every call to * is jsut a call |
| 13:38.44 | [TK]D-Fender | whether it's from a physical copper line from a telco, to a SIP phone, to an IAX soft-phone, to whatever |
| 13:39.26 | [TK]D-Fender | When I pick up the phone at my desk I can treat that just like a call from a line I have jacked into a card on the server |
| 13:47.48 | mtcdood_ | I don't understand what you mean at all |
| 13:48.04 | mtcdood_ | you need a device to connect to Asterisk, something |
| 13:48.26 | mtcdood_ | I was just asking about simulations because I wondered what the minimal investment was |
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| 13:50.29 | [TK]D-Fender | Yes and I just described using your own smart-phone which I presume you own of |
| 13:50.47 | mtcdood_ | how would my phone talk to Asterisk without some kind of card? |
| 13:50.55 | [TK]D-Fender | That talking to Asterisk is no different for what you asked about simulating or testing than a phsycial PCI card with telco lines jacked in |
| 13:51.03 | [TK]D-Fender | VoIP <- |
| 13:51.38 | [TK]D-Fender | Asterisk can use MANY different technologies |
| 13:51.54 | [TK]D-Fender | Hence its name |
| 13:53.05 | [TK]D-Fender | Make a voice menu. Grab my cell-phone with a SIP soft-phone app on it. Dial a number and have my server go to that menu I want to test. Done. |
| 13:53.45 | mtcdood_ | ok I'll look in that direction |
| 13:53.56 | mtcdood_ | don't know why I didn't think of VoIP |
| 13:54.10 | mtcdood_ | that's a pretty big oversight on my part |
| 13:55.01 | [TK]D-Fender | Yup... |
| 13:55.27 | [TK]D-Fender | How many people process calls on their server using ONLY cards with copper pluged into it? |
| 13:55.42 | [TK]D-Fender | Even those using it for copper lines probably have SIP phones. |
| 13:55.50 | [TK]D-Fender | So... those are calls just the same |
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| 16:27.45 | ruben23 | hi there guys need help my asterisk console seems only displaying white text on the console > the pink or red text are not appearing anymore |
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| 16:37.01 | Samot | OOOOO Oct 28th. Saving the date! |
| 16:40.22 | ruben23 | any idea guys.? |
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| 16:46.20 | Samot | ruben23: Did you do anything? |
| 16:46.28 | Samot | Make an update on the system? Update Asterisk? |
| 17:03.24 | qakhan | Hi all, I used AMI Park Action to park the calls and AMI ParkedCalls Action to get the list of parkedcalls. is there any why to pickup parked call user AMI? |
| 17:10.00 | Samot | The same way you get it out of the queue. |
| 17:10.02 | Samot | Redirect. |
| 17:10.09 | Samot | You redirect the call from X to Y |
| 17:24.35 | qakhan | Samot in Redirect i know the Channel. but i dont the Extra channel, which will pick the parked call. |
| 17:24.52 | Samot | It's an optional setting. |
| 17:25.15 | qakhan | i know the Exten which wants to pick up the parked call |
| 17:25.21 | Samot | OK |
| 17:25.35 | Samot | So the extra options are optional. |
| 17:25.39 | Samot | They aren't required. |
| 17:25.48 | Samot | You need the channel you're redirecting and the destination. |
| 17:25.56 | Samot | i.e. context,extension,priority |
| 17:26.56 | qakhan | but how to pass agent exten who wants to pictup parked call |
| 17:27.09 | qakhan | i am able to pass all the information |
| 17:27.12 | Samot | This is all logic in y our program. |
| 17:27.21 | Samot | How is the agent getting the call? |
| 17:27.30 | Samot | How will they grab this parked call? |
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