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| 00:21.22 | *** 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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| 12:22.31 | tsearle | hello all |
| 12:23.11 | file | moo |
| 12:23.16 | tsearle | oom! |
| 12:23.37 | tsearle | may I ask your advice ref codec negotiation? |
| 12:23.50 | file | sure |
| 12:24.05 | tsearle | we had a patch for ast11 / chan_sip to help reduce transcoding of asterisk |
| 12:24.20 | tsearle | so basically, |
| 12:24.34 | tsearle | A calls ast which forwards to B |
| 12:24.55 | tsearle | B sends an SDP answer back in a 183 or a 200 ok |
| 12:25.27 | tsearle | I would like to filter the codecs in the sdb answer send to A using the info of the SDP answer from B |
| 12:26.57 | tsearle | is there any way of pulling this off in pjsip? my co-worker who as been digging into it seems to indicate the sdp answer on the A leg gets pre-computed long before the answer from B arrives |
| 12:27.20 | file | the core doesn't convey the information sanely and there's no interface in 14 and below to do it |
| 12:27.30 | file | the stream work in 15 has the foundation to be able to do it |
| 12:27.39 | tsearle | ok |
| 12:28.06 | file | so, if you hacked up PJSIP and the core some in 13/14 it'd probably be possible... |
| 12:28.12 | tsearle | but if I were to put say a format_caps in the ANSWER control frame |
| 12:28.20 | file | theoretically yes. |
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| 12:29.13 | tsearle | so there isn't a limitation on pjsip preventing me from changing the media formats list at the time the answer is sent |
| 12:29.18 | tsearle | correct? |
| 12:29.55 | file | there's nothing at the PJMEDIA SDP level, it's the implementation and usage of it probably... that would need to change |
| 12:30.38 | tsearle | ok, well I'll take that as a promissing response and try to see if I have better luck than @ggo |
| 12:33.01 | tsearle | https://jenkins2.asterisk.org/job/check-ast-master-64/730/console |
| 12:33.07 | tsearle | I assume this is a false alarm? |
| 12:34.49 | file | yes |
| 12:42.46 | tsearle | hrm... still broke |
| 12:44.30 | file | blargh infrastructure |
| 12:45.02 | tsearle | I feel your pain |
| 13:07.39 | tzafrir | Regarding DAHTOOL-77 , how is the dahdi-complete tarball generated? From which files? |
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| 13:30.44 | tsearle | shit messed up |
| 13:31.37 | tsearle | commited my PJSIPDtmfMode fix with the commit id of the fast38 fix |
| 13:31.43 | tsearle | now I need to figure out how to revert |
| 13:32.09 | file | grab the old review and put it back up? |
| 13:33.05 | file | git review -d <review>,<patch set number> |
| 13:33.17 | file | will create a local branch with that review and patch set, then another "git review" to put it back up would do it |
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| 13:55.39 | file | attempts to find the misbehaving VM |
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| 14:04.39 | tsearle | I'm having an issue with gerrit |
| 14:04.50 | tsearle | I submitted a T.38 patch, then swtich to master |
| 14:05.02 | tsearle | do a hard reset to make sure everything is gone |
| 14:05.11 | tsearle | then I create a new branck for a new test |
| 14:05.24 | tsearle | but when it comes time to call get review |
| 14:05.32 | tsearle | it says I have multiple things to commit |
| 14:05.38 | tsearle | The outstanding commits are: |
| 14:05.38 | tsearle | b30d5f2 (HEAD -> ASTERISK-27085) testsuite: add a test for PJSIPDtmfMode |
| 14:05.38 | tsearle | e33323a (master) Merge branch 'master' of ssh://gerrit.asterisk.org:29418/testsuite |
| 14:05.38 | tsearle | 0af7476 res/res_pjsip_t38 add test to verify T.38 gets quickly rejected |
| 14:06.15 | file | are you sure you switched to master? it sort of seems as though you merged master into the current branch and then added another commit for a new test to the same branch |
| 14:06.34 | tsearle | tsearle@tsearle-Latitude-T440s:~/asterisk/testsuite$ git checkout master |
| 14:06.34 | tsearle | Switched to branch 'master' |
| 14:06.34 | tsearle | Your branch is ahead of 'origin/master' by 2 commits. |
| 14:06.34 | tsearle | <PROTECTED> |
| 14:06.34 | tsearle | tsearle@tsearle-Latitude-T440s:~/asterisk/testsuite$ git reset --hard master |
| 14:06.35 | tsearle | HEAD is now at e33323a Merge branch 'master' of ssh://gerrit.asterisk.org:29418/testsuite |
| 14:06.37 | tsearle | tsearle@tsearle-Latitude-T440s:~/asterisk/testsuite$ git pull |
| 14:06.39 | tsearle | X11 forwarding request failed on channel 0 |
| 14:06.41 | tsearle | Already up-to-date. |
| 14:06.56 | file | you seem to have altered your local master branch |
| 14:07.09 | tsearle | ok |
| 14:07.17 | tsearle | so then I should to a reset-hard to origin/master |
| 14:07.17 | file | if you do a git log do you see your T.38 test in there as well? |
| 14:07.49 | tsearle | yeah |
| 14:08.06 | file | then you somehow got that commit into your master branch |
| 14:08.12 | tsearle | ok |
| 14:08.24 | tsearle | so I did git reset origin/master instead of git reset master |
| 14:08.27 | tsearle | that looks a bit happer |
| 14:08.49 | tsearle | now I can delete and re-create the branch |
| 14:08.59 | file | git checkout -b ASTERISK-27085 origin/master |
| 14:08.59 | file | git cherry-pick b30d5f2 |
| 14:08.59 | file | git review |
| 14:09.13 | file | that'll create a new master branch off the origin one, cherry pick in only your test, and put it up for review |
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| 14:12.14 | tsearle | phew, thanks :-) |
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| 14:25.54 | seanbright | so if i have an m line in my sdp that has a dynamic payload number in it |
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| 14:26.03 | seanbright | but i don't have an a:rtpmap for that number, is that an error? |
| 14:29.04 | file | is the stream declined? |
| 14:29.30 | seanbright | pjsip bounces it |
| 14:29.37 | seanbright | chan_sip accepts it (surprise, surprise) |
| 14:30.06 | seanbright | sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP) |
| 14:30.18 | seanbright | this phone i am testing with sends 101 in the m line |
| 14:30.21 | seanbright | but no rtpmap for it |
| 14:30.47 | file | the RFC says SHOULD, which I think is silly |
| 14:31.15 | file | aha |
| 14:31.53 | file | If the payload type number is dynamically assigned by this session description, an additional "rtpmap" attribute MUST be included to specify the format name and parameters as defined by the media type registration for the payload format. |
| 14:33.33 | file | 101 is common for RFC2833... so it probably expects everyone to just know that |
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| 14:35.46 | file | it's cresl1n! |
| 14:36.00 | rmudgett | Who's that? |
| 14:38.09 | cresl1n | waves |
| 14:38.09 | seanbright | file: which RFC are you quoting? |
| 14:38.15 | seanbright | 4566? |
| 14:39.07 | seanbright | yes. yes it is. |
| 14:39.21 | seanbright | so the short version is that this phone is broked |
| 14:40.13 | file | ye ol' RFC2327 also sorta says it |
| 14:40.45 | file | For RTP formats using a dynamic |
| 14:40.45 | file | <PROTECTED> |
| 14:40.45 | file | <PROTECTED> |
| 14:40.45 | file | <PROTECTED> |
| 14:41.14 | seanbright | yeah. broke phone. |
| 14:42.46 | tsearle | waves |
| 15:23.18 | cresl1n | tsearle!!! |
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| 17:02.22 | tzafrir | I pushed branches called "next" to dahdi/linux and dahdi/tools in git.asterisk.org |
| 17:02.50 | tzafrir | They include various fixes that should probably be included in the next version. |
| 17:03.12 | tzafrir | However, it is not an RC: the branch is subject to rebasing |
| 17:03.41 | tzafrir | In case it is useful |
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| 17:47.15 | file | tests are good, tests find bugs |
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| 23:34.41 | snuff-work | seanbright: forgot that pjsip.conf.. now added with only relevant sip peers |