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| 14:18.26 | file | cresl1n, ! |
| 14:19.30 | cresl1n | file!!!! |
| 14:19.47 | file | cresl1n, today is Wednesday |
| 14:20.56 | cresl1n | Oh is it? |
| 14:21.03 | cresl1n | Well, that happened fast. |
| 14:22.11 | file | time flies |
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| 14:36.27 | file | definitely a Wednesday |
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| 15:41.08 | file | ooh |
| 15:41.10 | file | [Jun 29 11:11:39] DEBUG[16122] res_pjsip_session.c: Source of transaction state change is TRANSPORT_ERROR |
| 15:41.11 | file | neat. |
| 15:55.33 | cresl1n | Ah yes⦠the old "Source of transaction state change is TRANPORT_ERROR" trick... |
| 15:56.51 | cresl1n | If I had a dollar for every time I've seen that |
| 15:56.57 | cresl1n | file: :-P |
| 15:56.59 | gtjoseph | putnopvut: for the bridge tests, make sense to put them with the channel create and dial tests or put them under bridges? I was just going to rename the create_and_dial directory to create_dial_bridge then do 1 subdir for each. |
| 15:57.14 | tzafrir | Any chance sruffel or rmeyerriecks are around? |
| 15:57.19 | putnopvut | I like your plan gtjoseph |
| 15:57.23 | gtjoseph | cool |
| 15:57.25 | file | cresl1n, you'd have more if you came to Canada |
| 15:57.51 | cresl1n | But everything costs more too, right? |
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| 15:58.12 | file | shhhh |
| 15:58.52 | cresl1n | tzafrir: haven't seen sruffell around today |
| 16:00.49 | tzafrir | needs to know how I can move forward with https://issues.asterisk.org/jira/browse/DAHTOOL-74 |
| 16:01.12 | tzafrir | I'd like to push a package to Debian. |
| 16:01.49 | tzafrir | is currently at debconf, which will actually be in Canada next year |
| 16:16.48 | file | rmudgett, https://github.com/asterisk/asterisk/compare/master@%7B2016-06-9%7D...master@%7B2016-06-10%7D the 9th is success, the 10th is the first failure... for the goto_on_blindxfr test |
| 16:16.57 | file | but I don't understand what could have done it... |
| 16:16.59 | file | 'tis odd |
| 16:18.14 | coreyfarrell | file: about the codec.c ABI compat why do we want that in master? |
| 16:18.53 | file | coreyfarrell, imo if we can easily keep ABI for users of codecs then why not? also means that codec_g729a would work with master - may not be supported mind you, but if someone needed to then they could |
| 16:19.34 | gtjoseph | also makes any future work in codec easier to cherry-pick |
| 16:19.43 | gtjoseph | from 13 to master |
| 16:21.34 | rmudgett | file: You talking about the date? |
| 16:21.58 | file | rmudgett, yes - I looked back in Jenkins and the first failure started on the 10th, it was 100% passing before then |
| 16:22.50 | file | looking at the actual cause of the failure it seems as though the UDP socket returned EPERM |
| 16:23.21 | file | how bizarre |
| 16:23.32 | file | is now playing. |
| 16:23.54 | file | it's also only this test... |
| 16:24.37 | file | goes to look at other transfer failures |
| 16:24.58 | coreyfarrell | file / gtjoseph: thanks, just felt the need to ask since it causes an extra struct alloc |
| 16:25.19 | gtjoseph | no problem |
| 16:25.48 | rmudgett | What extra struct alloc? There is no extra alloc. |
| 16:26.36 | coreyfarrell | rmudgett: thanks, I see that now too :) |
| 16:33.03 | file | moves "Switch to Elementary OS" higher on his list as he receives a barrage of late notifications |
| 16:39.46 | file | gtjoseph, Alexei returneth |
| 16:39.59 | gtjoseph | <sigh> |
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| 17:54.53 | gtjoseph | putnopvut: got an issue with StasisEnd (or lack thereof)... I dial a local channel which plays 3 sec silence and hangs up. I hang up immediately after dialing. I get my own ChannelHangupRequest, then 3 seconds later get the dialplan's ChannelHangupRequest but I never get a StasisEnd and the channel still appears when I get the channel list. |
| 17:54.59 | gtjoseph | Something I'm missing? |
| 17:55.21 | putnopvut | Hm |
| 17:55.58 | putnopvut | I don't think you're missing anything. Sounds like a bug to me. |
| 17:56.01 | gtjoseph | i've tried changeing the order so the dialplan hangs up first but no change is results |
| 17:56.19 | gtjoseph | :) |
| 18:26.14 | seanbright | does https://gerrit.asterisk.org/3100 need another +1 review to get automagically merged? |
| 18:26.33 | file | needs someone to +2 it now |
| 18:26.37 | file | and I just did |
| 18:26.38 | seanbright | ah |
| 18:26.43 | seanbright | file: can you +2 it now? |
| 18:26.44 | seanbright | nice |
| 18:36.54 | putnopvut | SILK woes: http://paste.fedoraproject.org/386439/14672253/ |
| 18:37.38 | putnopvut | I'm curious if the translation paths in what I just pasted look wrong to anyone. I'm trying to figure out whether there's a bug in decoding SILK audio or if there's a bug in setting up the translation path (or both) |
| 18:38.18 | file | reads |
| 18:39.06 | file | yes, I think that should be slin@16 not slin@8 |
| 18:39.38 | file | or if it were slin@8 then the read transcode on jitsi should be to slin@8 |
| 18:39.49 | file | but really, it should be 16 end to end |
| 18:39.58 | putnopvut | Yeah, 16 end-to-end is what I would have expected. |
| 18:40.04 | putnopvut | Not sure why slin8 is there. |
| 18:40.37 | putnopvut | I'll try to think of a better way of isolating SILK decodes. |
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| 19:18.40 | file | gtjoseph, I'm a bit confused over why Alexei's testsuite changes work on un-patched |
| 19:18.54 | gtjoseph | you and me both |
| 19:19.12 | gtjoseph | won't have time to look until later |
| 19:20.37 | gtjoseph | one of the tests i can understand because it just adds another state to the regex |
| 19:20.53 | gtjoseph | the other I thought would fail for sure |
| 19:21.21 | gtjoseph | i have to admit though that i'm bad at reading the comment history. :) |
| 19:28.58 | gtjoseph | putnopvut: on the dial test plan, are you expecting the channel to be answered when the invalid ops tests run? |
| 19:29.48 | putnopvut | No, it should not be answered |
| 19:29.58 | gtjoseph | good. |
| 19:30.00 | gtjoseph | :) |
| 19:30.43 | gtjoseph | otherwise the tests all return 409 if answered but not in stasis and succeed if both |
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| 20:18.45 | gtjoseph | putnopvut: issue with continue as well. the continue works but the only ari event issued after the continue is a StasisEnd where I should be seeing ChannelDialplan events as in the originate_then_continue test. |
| 20:19.36 | putnopvut | That's certainly interesting. I'm not sure why that behavior would be different... |
| 20:19.59 | gtjoseph | me neither. |
| 20:20.41 | gtjoseph | should i even be seeing a StasisEnd after the continue? |
| 20:23.11 | putnopvut | I mean, I think so? The channel is exiting Stasis to execute dialplan code. |
| 20:24.29 | gtjoseph | but if i get the StasisEnd should I even expect any more events then? |
| 20:25.00 | putnopvut | If you get a StasisEnd, then I think that has an implicit "unsubscribe" with it as well, so you wouldn't get further channel events. |
| 20:25.45 | gtjoseph | The current originate_then_continue DOES get events after that which is weird then unless it's something goofey the ari originate test case is doing. |
| 20:26.57 | gtjoseph | mayube i just need to put a call to Stasis in the target extension |
| 20:36.24 | gtjoseph | well, i get a new StasisStart event but still nothing after that. |
| 20:37.00 | gtjoseph | I think i'm going to write the tests as I think they should work based on the existing tests, then open issue for the ones that don't work. |
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| 20:48.14 | file | never give up, never surrender! |
| 20:54.52 | seanbright | does one need special permissions to review something in gerrit? |
| 20:55.30 | file | you need to have a valid license agreement on file |
| 20:55.31 | seanbright | nm |
| 20:55.33 | seanbright | found it |
| 20:55.58 | seanbright | i helped! |
| 20:55.58 | putnopvut | I figured out why my translation paths were being built incorrectly. It's because I was using the wrong format name on my silk->slin translators. |
| 20:56.12 | file | putnopvut, oic |
| 20:56.15 | putnopvut | So now I have SILK translating correctly in both directions. |
| 20:56.18 | putnopvut | yee-ha |
| 20:56.49 | file | putnopvut, yay |
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