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| 00:19.00 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! blogs.asterisk.org -=- Follow on Twitter at @AsteriskDev |
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| 14:46.04 | file | gtjoseph: https://jenkins2.asterisk.org/job/gate-testsuite/1/console |
| 14:46.05 | file | much sad |
| 14:46.23 | gtjoseph | Ha! |
| 14:46.50 | gtjoseph | I keep forgetting that script approval is turned on |
| 14:47.04 | gtjoseph | any admin can do it. it's under manage jenkins |
| 14:47.28 | gtjoseph | done |
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| 14:49.03 | gtjoseph | nad i typod something. |
| 14:53.56 | rmudgett | i typed something too |
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| 18:23.24 | stefan27 | putnopvut == rmudgett ? |
| 18:23.41 | rmudgett | no |
| 18:23.46 | rmudgett | rmudgett == rmudgett |
| 18:25.07 | stefan27 | ah sorry |
| 18:26.33 | stefan27 | but putnopvut does use IRC once in a while? |
| 18:28.39 | stefan27 | basically I'm trying to guess whether an asterisk developer will look at ASTERISK-26982 within some weeks or if it's worth it for me to look into the code |
| 18:29.54 | stefan27 | but we have a bad habit of patching in our own "work-arounds" for bugs that later conflict with the trunk |
| 18:30.01 | file | he does, he's just gone early today |
| 18:30.04 | stefan27 | happend with DTLS and SRTCP |
| 18:30.19 | file | I'd expect it to get looked at in a few weeks |
| 18:30.31 | file | probably 2 |
| 18:30.38 | file | maybe sooner |
| 18:30.53 | stefan27 | cool it's not like i'm trying to scream loud for attention to make it seem more important |
| 18:31.21 | stefan27 | for us I dont think it matters that much if it's fixed before chrome 59 |
| 18:31.40 | file | has anyone tested PJSIP? |
| 18:33.24 | stefan27 | no, but I can make a short note in the jira that it may not be just chan_sip |
| 18:36.13 | file | that's not needed, I was more hoping for a more concrete yes or no but *shrug* |
| 18:36.48 | stefan27 | anyone with a firefox browser can test it via SIPml demo |
| 18:36.55 | stefan27 | if they still host that call demo |
| 18:44.24 | stefan27 | scgm11 that tested uses chan_sip too, but I've talked to other webrtc people that use pjsip, forgot their names though |
| 18:57.06 | stefan27 | time for weekend, byebye! |
| 18:58.23 | file | cheers |
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| 22:09.04 | rmudgett | file: Is there provisions for stream topologies to have declined/disabled in the code? |
| 22:09.19 | rmudgett | Would that be the UNKNOWN type? |
| 22:09.42 | file | it could be |
| 22:09.52 | rmudgett | Or is the format cap empty? |
| 22:09.59 | file | you could do an unknown stream type, empty format cap, and removed as state |
| 22:10.08 | file | and it'd be an empty placeholder of nothing |
| 22:10.32 | file | we could also add an explicit declined state |
| 22:11.04 | rmudgett | I think the removed state would be better that way we won't lose the audio/video stream type. |
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