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| 14:19.43 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Hell hath frozen over: Asterisk is moving to Git -=- Follow on Twitter at @AsteriskDev |
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| 14:30.28 | file | additional dependencies installed |
| 14:32.36 | mjordan | sighs |
| 14:32.47 | mjordan | common allocation routine be damned |
| 14:42.04 | *** join/#asterisk-dev rmudgett (rmudgett@nat/digium/x-fntxyjmrsnsizype) |
| 14:53.12 | mjordan | file: Let's see if automerge works for Traud's patch on 11 |
| 14:53.19 | mjordan | I'll +2 it and let it do its gating thing |
| 14:53.27 | file | doesn't hurt to try |
| 14:54.00 | mjordan | well, if only the pjsip channel driver tests are failing... |
| 14:54.28 | mjordan | And assuming my "ignore the result when there are no tests" works as well... |
| 14:54.34 | mjordan | Then it might just work |
| 14:58.43 | mjordan | drat |
| 14:58.44 | mjordan | 2015-09-28 09:57:47 hudson.AbortException: None of the test reports contained any result |
| 14:58.52 | mjordan | I'll have to look into that again. |
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| 15:22.06 | mjordan | drat |
| 15:22.16 | mjordan | the junit test results will always fail a test if it contains no results |
| 15:22.17 | mjordan | Hm. |
| 15:22.25 | file | aw |
| 15:24.42 | mjordan | Only thing I can think to do is to separate out the ARI/External MWI plans |
| 15:25.00 | mjordan | or else hack the scripts to check the branch and bail |
| 15:25.15 | mjordan | but most of this stuff is parameter driven, and the scripts don't really "know" that they are the ARI/ExternalMWI plan |
| 15:25.30 | mjordan | ponders |
| 15:37.21 | mjordan | looks like the TCP auth limit stuff still doesn't work either |
| 15:37.21 | mjordan | https://jenkins.asterisk.org/jenkins/job/gate-asterisk-channel-drivers/75/ |
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| 16:06.33 | superscrat | leifmadsen, I started reading that book finally. |
| 16:06.51 | superscrat | coreyfarrell, you appear! |
| 16:09.11 | file | superscrat, https://jenkins.asterisk.org/jenkins/job/gate-asterisk-channel-drivers/75/ look what's failing :D |
| 16:23.08 | superscrat | oh, awesome. |
| 16:23.38 | superscrat | file, at least it's polite with its message. |
| 16:27.32 | superscrat | file, how do i get the testsuite and asterisk logs from jenkins? |
| 16:27.59 | file | download the logs under build artifacts? |
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| 16:51.05 | superscrat | file, blah, right. I had already clicked off of that view and I guess was expecting to see it as an option on the side. Thanks. |
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| 18:49.06 | putnopvut | I'm having symbol errors and it's annoying. I had Asterisk 13 compiled with MALLOC_DEBUG, and when I attempt to connect an ARI client to Asterisk, Asterisk shuts down with: "*CLI> asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_ari.so: undefined symbol: ast_websocket_uri_cb" |
| 18:49.37 | putnopvut | So when I make distclean and recompile without MALLOC_DEBUG, then I can't even get Asterisk started because "asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_ari.so: undefined symbol: __ast_strdup" |
| 18:50.04 | putnopvut | So my question is: wtf |
| 18:50.31 | file | I assume you wiped the modules directory? what's the timestamp on stuff? |
| 18:50.58 | putnopvut | I wiped. In my latest build, I re-enabled MALLOC_DEBUG. Let me check timestamps |
| 18:51.39 | putnopvut | The timestamps on all modules are from 7 minutes ago |
| 18:52.38 | file | interesting |
| 18:53.08 | file | did a distclean too? |
| 18:58.29 | putnopvut | Yep, that's what I did when disabling MALLOC_DEBUG |
| 18:58.44 | putnopvut | I'll do another just so I don't have MALLOC_DEBUG enabled for now. |
| 19:23.39 | putnopvut | Just tried an experiment: Turned on DONT_OPTIMIZE, and turned off MALLOC_DEBUG, so I'd have a res_ari.i file to inspect. In it, I don't see any reference at all to __ast_strdup(). However, when I `nm res/res_ari.so | grep strdup` I do see a reference to __ast_strdup() in there. |
| 19:23.53 | file | huh |
| 19:24.02 | file | what happens if you manually remove res/res_ari.so and make? |
| 19:24.26 | putnopvut | Just did that and it didn't help |
| 19:24.59 | rmudgett | putnopvut: Try removing the installed libasteriskssl.so |
| 19:25.50 | putnopvut | Okay, just did that and now I'm doing a complete re-build from scratch |
| 19:31.55 | putnopvut | Didn't help |
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| 19:56.32 | putnopvut | So, making a new clone of Asterisk, it works fine. Doing make uninstall, make distclean, and rebuilding everything again on my "tainted" version results in the same thing still occurring. |
| 19:56.38 | putnopvut | I have no idea what is going on here. |
| 19:57.08 | file | UNCLEAN! UNCLEAN! BURN IT |
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| 20:01.23 | X-Rob | file: you around? |
| 20:01.30 | file | yes |
| 20:01.52 | X-Rob | So I've got a guy in #freepbx-dev who's ACTUALLY got a device that works with ipv6 - apparently a snom m9 |
| 20:02.06 | rmudgett | he is square not round |
| 20:02.11 | TazzNZ | I know file, he knows me :) |
| 20:02.12 | X-Rob | Is there any other endpoints that you know of that do ipv6? |
| 20:02.23 | file | I believe Bria does |
| 20:02.26 | file | for a softphone |
| 20:02.40 | file | hardphone wise... nothing endpointy comes to mind |
| 20:02.46 | X-Rob | Bah. |
| 20:02.56 | X-Rob | I'm going to be whinging about this to everyone as astricon |
| 20:03.13 | X-Rob | Anyway, so apparently pjsip won't work without an explicit ipv6 transport, what about chansip? |
| 20:03.27 | file | no clue |
| 20:03.32 | X-Rob | Hah |
| 20:03.32 | X-Rob | <3 |
| 20:03.46 | X-Rob | I had a suspicion you were gunna say that |
| 20:03.52 | X-Rob | TazzNZ: Soooo, it's all up to us. |
| 20:04.11 | X-Rob | Thanks anyway, file. We shall piss off and investigate. |
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| 20:10.54 | WIMPy | Have you ever noticed that at the en of a call, rtpaudioqos is set twice? |
| 20:11.49 | WIMPy | The other rtp* variables are not. |
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| 20:22.08 | putnopvut | Finally got myself into a working state. Had to just nuke my 13 branch and recreate it, but I'm up and running. |
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| 21:10.05 | jkroon | <PROTECTED> |
| 21:10.05 | jkroon | we can force a switch to t38 (which based on my testing should work but obviously the detection needs to work first). This is in res/res_fax.c (which uses framehooks to get the raw frames) |
| 21:15.57 | jkroon | I've also looked at the MixMonitor code, but that uses audiohooks not framehooks, and it seems the audiohooks themselves already performs transcoding. |
| 21:16.49 | jkroon | [Sep 28 23:02:24] DEBUG[16904][C-00000003] channel.c: Set channel SIP/IS-00000006 to read format slin |
| 21:16.49 | jkroon | [Sep 28 23:02:24] DEBUG[16904][C-00000003] res_fax.c: Attached FAX detect to channel SIP/IS-00000006. |
| 21:17.51 | jkroon | so that does seem to set the frame read format to slin - but the framehook is still receiving g729 - is there thus any point in changing the read format for the channel - doesn't that simply (temporarily) cause a pointless transcode to happen? |
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| 23:27.56 | t4nk997 | Hi, my team and I have created a real-time payments platform for asterisk - any shrewd developers interested in trying it? |
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