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| 03:13.37 | jpsharp | So I'm having an issue with an ODBC connection. Despite having "pre-connect => yes" and "idlecheck" set 15, i'm always getting database timeouts and reconnects. And the reconnects sometimes take 8-10 seconds, which makes my dialplan application act dead at times. Suggestions on how to troubleshoot/fix this? |
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| 03:47.45 | overyander | jpsharp what does your topology look like? are the ast and db servers separated by WAN or something? |
| 04:10.09 | jpsharp | No, they're on the same box. I'm talking to localhost over a unix socket. |
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| 04:23.12 | overyander | are you positive it's over socket and not loopback or LAN IP? |
| 04:23.40 | overyander | if so, then i would start watching top to see what's going on when it hangs |
| 04:24.02 | overyander | what else is using that DB? |
| 04:30.01 | jpsharp | Nothing else is using the db. the box is 99% idle. It's just a dev machine. |
| 04:30.44 | jpsharp | And from my understanding of the mysql api, if you specify localhost, it will always attempt to use a unix domain socket unless you tell it not to. |
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| 04:46.47 | overyander | jpsharp are you using a custom function that requires db lookups or are you trying to use realtime/db based dialplan? |
| 04:46.59 | jpsharp | ODBC function. |
| 04:47.42 | jpsharp | Oddly enough I seem to have fixed it by changing the idletimeout to 3600. Maybe I don't quite understand what the idltimeout does. |
| 04:52.45 | overyander | glad you got it sorted |
| 04:57.19 | jpsharp | Yeah, now I just have to figure out how to properly set variables while doing a GoSub(). |
| 04:57.45 | jpsharp | You'd think I'd know this stuff after playing with Asterisk for damn near 15 years. |
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| 10:08.42 | Dovid | what is NBScat used for ? |
| 10:08.55 | Dovid | the documentation does not have much on it |
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| 12:51.17 | donnib | Hi |
| 12:51.17 | donnib | i am trying to get a softphone connected to asterisk, the softphone is connecting to the network with VPN (openvpn) and get's an ip 172.17.0.1/16 and my internal network is 10.10.0.1/24 |
| 12:51.24 | donnib | when i do sip show peers i can see the softphone has the ip 10.10.0.1 which is my router (also where the vpn server is) |
| 12:51.32 | donnib | if reregister the softphone i can see that for a quick moment it registers with the right ip 172.... but then changes to 10.10.0.1 |
| 12:51.40 | donnib | how can i overcome this problem ? |
| 12:59.18 | [TK]D-Fender | Show us |
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| 13:01.44 | donnib | for some reason it seems like the problem is solved now, i jus removed and created the extension and it seems like the sound goes thru now |
| 13:24.42 | keithf | donnib: do you have NAT set to yes on the extension? |
| 13:45.46 | *** join/#asterisk skirmisha (~skirmisha@213.16.62.70) |
| 13:45.52 | skirmisha | hi guys |
| 13:46.31 | skirmisha | on asterisk ver 13 how can i force playpack function to not search for sln format, file is already in .wav format, but asterisk is searching for sln format and thus it fails |
| 13:47.03 | [TK]D-Fender | And why would it fail? |
| 13:47.09 | [TK]D-Fender | it would transcode like normal |
| 13:47.18 | [TK]D-Fender | And you can't tell it to specifically NOT look for 1 format |
| 13:47.28 | [TK]D-Fender | it'll pick the most efficient version available |
| 13:48.01 | skirmisha | it is requesting that file extension whatever i do, but that file does not exist, only .wav extension exist |
| 13:48.34 | [TK]D-Fender | You don't specify an extension |
| 13:49.22 | skirmisha | no i did not |
| 13:49.26 | skirmisha | Playback failed |
| 13:49.43 | skirmisha | app_playback.c:493 playback_exec: Playback failed on SIP |
| 13:50.17 | [TK]D-Fender | Show in full |
| 13:50.25 | skirmisha | Playing 'custom/example.slin' (language 'en') |
| 13:50.30 | [TK]D-Fender | Along with the folder dump |
| 13:50.35 | [TK]D-Fender | in full |
| 13:52.52 | skirmisha | dont have folder dump, but let me try something |
| 13:58.15 | skirmisha | <PROTECTED> |
| 13:58.15 | skirmisha | <PROTECTED> |
| 13:58.15 | skirmisha | <PROTECTED> |
| 13:58.16 | skirmisha | --WARNING[5234][C-00008291]: app_playback.c:493 playback_exec: Playback failed on SIP/test-0000f249 for custom/example,noanswer |
| 13:59.58 | skirmisha | and the file that is in directory is example.wav |
| 14:00.07 | skirmisha | slin does not exist at all |
| 14:00.30 | [TK]D-Fender | Show us |
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| 14:01.10 | skirmisha | is there a command to debug playback function? |
| 14:05.13 | [TK]D-Fender | core debug is all we've got |
| 14:05.16 | [TK]D-Fender | not jsut verbose |
| 14:09.09 | skirmisha | let me increase it |
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| 14:36.50 | donnib | if i set allow=opus and my trunk get's registered, does that mean the provider supports it ? |
| 14:37.24 | donnib | when i get a call i get no compatible codecs....... but i am unsure whether that is the provider or me that has a problem |
| 14:37.30 | [TK]D-Fender | no |
| 14:37.35 | [TK]D-Fender | Tho proves notthing |
| 14:37.41 | [TK]D-Fender | that's saying * will offer it |
| 14:37.54 | [TK]D-Fender | It has nothing to do with registration at all, and doesn't mean they'll accept it for a call |
| 14:38.03 | [TK]D-Fender | And you have to actually look at what is being negotiated |
| 14:38.19 | [TK]D-Fender | You should also already have the list fo what the provider supports from their site as it is |
| 14:38.31 | donnib | ok so i get SIP/2.0 488 Not acceptable here |
| 14:38.43 | [TK]D-Fender | And I've never heard of ANY provider supporting it |
| 14:39.46 | donnib | oh ok, yes true i can see it's not on the list |
| 14:39.53 | donnib | they do however support speex |
| 14:40.08 | donnib | let me try that |
| 14:40.29 | donnib | what's the difference between speex, speex16 and speex32 ? |
| 14:40.39 | donnib | bandwidth ? |
| 14:40.52 | donnib | and what is speex then using of bandwidth |
| 14:41.31 | [TK]D-Fender | https://www.google.ca/#q=asterisk+speex+bandwidth |
| 14:41.38 | [TK]D-Fender | All of this is readily Google-able |
| 14:42.59 | donnib | hmm changed to allow=speex and i still get SIP/2.0 488 Not acceptable here |
| 14:43.52 | [TK]D-Fender | And you're not showing the full call or their page saying what they support |
| 14:43.58 | [TK]D-Fender | and that is still not one of the standards |
| 14:44.07 | [TK]D-Fender | You should be sticking tto the normal ones. |
| 14:44.13 | [TK]D-Fender | G.729, G.711 |
| 14:44.20 | [TK]D-Fender | As your system supports |
| 14:44.36 | donnib | ok, i thought they were old and the newer ones gave better quality |
| 14:44.41 | [TK]D-Fender | These oddball codecs are not what provider's normally support |
| 14:45.08 | donnib | https://www.fonet.dk/#Gen_8 |
| 14:47.51 | [TK]D-Fender | [10:43][TK]D-FenderAnd you're not showing the full call |
| 14:51.12 | donnib | how much should i anonymize ? |
| 14:51.30 | [TK]D-Fender | Don't screw with the evidence |
| 14:51.54 | donnib | well no but there are all sorts of info in the log like ip addresses and, usernames etc |
| 14:52.13 | [TK]D-Fender | Leave the IP's |
| 14:52.22 | [TK]D-Fender | we need proof of what's offered |
| 14:52.26 | [TK]D-Fender | the username doesn't matter |
| 14:53.19 | donnib | https://gist.github.com/donnib/04c071533a405fcfbe35 |
| 14:54.10 | [TK]D-Fender | Where's your config to match? |
| 14:54.28 | donnib | ouch...i use freepbx |
| 14:55.03 | [TK]D-Fender | And? |
| 14:55.15 | [TK]D-Fender | The PEER CONFIG is a dumb text block |
| 14:55.21 | [TK]D-Fender | COPY/PASTE |
| 14:55.26 | [TK]D-Fender | mask the secret |
| 14:55.48 | [TK]D-Fender | And their invite does NOT include Speex |
| 14:55.50 | donnib | this ? https://gist.github.com/donnib/7f5f8a92248d7fd5024e |
| 14:56.03 | [TK]D-Fender | And their invite does NOT include Speex |
| 14:56.04 | [TK]D-Fender | ^^^ |
| 14:56.13 | donnib | hmm well then that's it, on thei website they state Speex |
| 14:56.18 | [TK]D-Fender | Line 62-66 |
| 14:56.21 | [TK]D-Fender | it is NOT being offered |
| 14:56.25 | [TK]D-Fender | so make a valid choice |
| 14:57.30 | donnib | thank you |
| 15:01.09 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 15:02.03 | bounceman | Any idea how to troubleshoot dtmf issues? We have a minority of calls in wich DTMF is not recognised. Analyzing the call recording you can clearly hear it... however I just want to exclude some devices on the network. |
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| 15:57.33 | wdoekes | ~voip-info |
| 15:57.33 | infobot | methinks voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
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| 16:00.07 | [TK]D-Fender | Dead WIKI |
| 16:00.09 | [TK]D-Fender | Do nott use |
| 16:00.25 | [TK]D-Fender | (or as last resort |
| 16:00.32 | Qwell | infobot: no, voip-info is dead. Don't use it. |
| 16:00.32 | infobot | Qwell: okay |
| 16:00.36 | Qwell | ~voip-info |
| 16:00.36 | infobot | hmm... voip-info is dead. Don't use it. |
| 16:05.18 | bounceman | I think the Asterisk wiki is really bad... but hey that is just me. |
| 16:05.38 | [TK]D-Fender | Astterisk |
| 16:05.53 | [TK]D-Fender | Asterisk's Wiki is alive and well. Voip-info is not |
| 16:06.15 | file | the Asterisk wiki is most definitely not dead |
| 16:08.28 | bounceman | I do not said it was dead, I just think it has very little info for a platform that has been around since early 2000s. Of course it has changed a lot over the years. |
| 16:08.38 | bounceman | I am not contributing by the way o.O |
| 16:08.43 | file | what's missing? |
| 16:09.07 | file | I think we have a lot of stuff - but Confluence searching and navigation presents a problem for finding it |
| 16:09.18 | bounceman | I cannot give you anything right here, I've just encountered wiki sites over the last year that has very little information. That is all |
| 16:09.36 | bounceman | wiki sites meaning sites on the asterisk wiki |
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| 16:11.19 | file | do you mean wiki pages, or sites linked off the wiki? |
| 16:11.23 | bounceman | https://wiki.asterisk.org/wiki/display/AST/Channel+Variables |
| 16:11.25 | bounceman | This for an example |
| 16:12.19 | file | if you click the down arrow beside the page on the left there's more pages |
| 16:12.22 | [TK]D-Fender | What about it? |
| 16:12.29 | bounceman | I just think it is badly organised. |
| 16:12.37 | bounceman | Now I cannot be with you any more I have to catch a train |
| 16:12.54 | bounceman | And it could indeed contain better information without forcing the user into sublevels |
| 16:12.58 | [TK]D-Fender | Well you haven't suggested what you think is bad.. or what to replace it with |
| 16:13.18 | [TK]D-Fender | So until then... we still seem to do just fine with it. When you have something to suggest let us know |
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| 18:34.42 | lvlinux | hey guys, what's the best way to play a sound to only one side of a call? |
| 18:35.33 | lvlinux | i.e. one side is hearing ringing or something else, and the other side is hearing "press 1 to accept this call from blah blah..." or somesuch. |
| 18:36.05 | lvlinux | at the beginning of a call, not in the middle. |
| 18:36.23 | lvlinux | Local channels? Or is there something better suited? |
| 18:40.07 | [TK]D-Fender | "core show application dial" <- |
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| 18:42.55 | lvlinux | [TK]D-Fender: oh ok sweet. I didn't think dial would do it so I didn't even check that... |
| 18:43.01 | lvlinux | thx |
| 18:43.29 | [TK]D-Fender | VERY FINE MANUALS! |
| 18:43.31 | [TK]D-Fender | :) |
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| 19:40.20 | EllenorL | hi |
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