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| 07:54.17 | bombev | good morning to all |
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| 13:07.25 | jkroon | hi all |
| 13:08.01 | jkroon | file, do you have a minute or 10 for me regarding the channel owners for rtp_instance issue from yesterday? |
| 13:08.15 | file | I just awoke but sure |
| 13:09.10 | jkroon | lol ok |
| 13:10.07 | jkroon | i submitted a patch to https://issues.asterisk.org/jira/browse/ASTERISK-20754 - it doesn't work, but it seems at the point where I call the new ast_rtp_instance_set_chan function the owner member of the sip_pvt structure is NULL - which doesn't make sense for me ... ideas/suggestions? |
| 13:10.17 | file | at the time of RTP instance creation there probably isn't an owner, it gets created later probably |
| 13:10.46 | file | the owner can also change, which is updated using the fixup callback |
| 13:10.52 | jkroon | ok, so we have a dialog, with an rtp instance, but no channel yet? |
| 13:11.00 | file | correct |
| 13:11.04 | jkroon | ok, so I'm barking up the wrong tree ? |
| 13:11.29 | file | well, you can't just set it there... so yes? |
| 13:11.50 | file | find where the owner is allocated, and also the fixup callback where the owner can change... those are the two places off the top of my head |
| 13:11.59 | jkroon | crap. is there any way to (directly or indirectly) track down the channel (if any) associated with an rtp_instance? |
| 13:12.22 | file | owner can also become NULL, at which point you would want to set it to NULL in the instance |
| 13:12.25 | jkroon | ok, let me try that first, for now if it only works for SIP that's good enough. |
| 13:12.37 | jkroon | ok, so right idea, wrong place. |
| 13:12.55 | file | no - the fact they use an RTP instance is an implementation detail not exposed so you can't track down the channel that belongs to an rtp_instance outside of adding it like you are doing |
| 13:13.34 | jkroon | ok, in dialog_initialize_rtp - will dialog->owner *always* be NULL, or does that again depend? |
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| 13:13.50 | file | I'd have to look at many places in chan_sip to answer that |
| 13:13.59 | acidfoo | back from vacation :} |
| 13:14.04 | file | off the top of my head I'm pretty sure |
| 13:14.06 | jkroon | ok, so i'm going to assume that it can be non-null and just call _set_chan anyway :) |
| 13:19.06 | jkroon | ok, so sip_new seems to be the most sensible place ... gets passed a sip_pvt which has pointers to the rtp_instances and allocated the new ast_channel, so just let the rtp_intance know about the new channel. You mentioned something about a callback? |
| 13:20.45 | file | fixup |
| 13:21.04 | jkroon | sip_fixup ... ok, i'm not sure how to change that particular function to be honest. |
| 13:21.30 | file | it changes the owner, you have to change it in the rtp instance too or it'll be the wrong channel |
| 13:21.41 | jkroon | just update the channel for p->{rtp,vrtp,trtp} to newchan? |
| 13:21.56 | file | yes |
| 13:22.15 | file | and in the hangup callback the owner will go away and should also go away in the RTP instance |
| 13:23.06 | jkroon | agreed. |
| 13:23.16 | jkroon | thanks for giving me that location - was about to ask. |
| 13:23.51 | jkroon | dialog_unlink_all too? |
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| 13:24.40 | file | ummm... wherever it nulls the owner |
| 13:27.13 | jkroon | ok, so sip_hangup and dialog_unlink_all both :) |
| 13:29.21 | file | FRIDAY! that is all. |
| 13:30.12 | jkroon | well, if this is done and I can get some kind of OK on it ... guess what ... WEEKEND! |
| 13:32.10 | jkroon | wondering whether I should abstract the updating of that owner into a sip_dialog_set_owner() which updates ->owner and makes the required ast_rtp_instance_set_chan() calls and then use that everywhere instead ... |
| 13:33.42 | file | that sounds like a fine idea |
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| 13:35.53 | jkroon | cool, let me just test this with the code littered all over then I'll clean up and submit an updated patch. |
| 13:36.12 | jkroon | what do I do with non-SIP channels that I can't test? I can guess at a patch but I cannot test them. |
| 13:37.04 | file | then go for that and ask others to test |
| 13:39.40 | jkroon | one big patch or a separate patch per technology? |
| 13:40.16 | file | the changes overall should not be that huge, one patch is fine |
| 13:40.30 | jkroon | ok, will split it to -tested and -untested at least then. |
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| 13:44.13 | wdoekes | jkroon: you'll probably want a single reviewboard item => just one patch |
| 13:44.55 | jkroon | wdoekes, ok, but I'll still make it clear that I only ever tested SIP. |
| 13:52.30 | jkroon | great success! file - thanks for your assistance, just patching the other channel types too then I'll submit - where can I find people to test? |
| 13:53.43 | file | mailing list, IRC, places! |
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| 13:57.20 | jkroon | hmm, here is an interesting one, jingle_pvt has a vrtp member, but it's not set anywhere, ever, it's however tested for being set in a few places. |
| 13:57.58 | file | chan_jingle has no video support |
| 13:58.27 | jkroon | because no one added it yet or because the protocol doesn't support it? |
| 13:59.06 | file | chan_jingle is deprecated and no longer maintained |
| 13:59.14 | file | it has been replaced with chan_motif which does have video support |
| 14:01.20 | file | as well - all new features/functionality are done against trunk |
| 14:02.59 | jkroon | oh crap, been building patch against 1.8.18.0 ... :( |
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| 14:51.55 | jkroon | file, ok, uploaded a patch against 1.8.18.0 - will do a rebase against trunk later, but please do check if conceptually the idea is correct thank you very much. |
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| 15:22.06 | elguero | if anyone gets bored (or wants to be bored, depends on how you view code reviews), this one has been up on reviewboard for a little bit, https://reviewboard.asterisk.org/r/2068/... thanks |
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| 15:28.17 | WIMPy | I'd go with file size. |
| 15:29.41 | jsmith | putnopvut: Over 256,000 calls with no more of those crashes we were seeing. If you haven't yet pushed that patch, please do so :-) |
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| 15:32.50 | putnopvut | jsmith: that's good enough for me. I'll go ahead and commit. Is it okay to close the corresponding issue, too? |
| 15:33.03 | jsmith | putnopvut: Please :-) |
| 15:33.09 | putnopvut | Okay, will do. |
| 15:33.12 | jsmith | putnopvut: I can always re-open it later, right? |
| 15:33.24 | elguero | WIMPy: What do you mean? If you mean go by if the file size is 0, that doesn't work because we create the file with a header even if there is no audio written to it. |
| 15:34.31 | WIMPy | elguero: Sure, but you can test for a limit. |
| 15:37.52 | elguero | WIMPy: hmm... would that work with different file formats? Would the headers be different sizes? |
| 15:39.00 | WIMPy | Possibly. But does it write any headers other than for wav? |
| 15:40.31 | WIMPy | But it should be possible to find a value that's bigger than any header but smaller than any sensible amount of audio. something like 128-1024 bytes. |
| 15:40.56 | elguero | You can create files in other formats besides wav, so, I would say yes. |
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| 15:41.45 | elguero | Well, I was trying to go with an accurate approach... that way you know for sure if audio was written to the file rather than guessing and accidentaly erasing a file that should not be erased |
| 15:41.45 | WIMPy | I'm not sure any others have headers. |
| 15:42.46 | WIMPy | That gives options for more features ;-) |
| 15:43.09 | WIMPy | If you count frames ou could have a configurable minimum length in seconds. |
| 15:43.55 | WIMPy | puts the visions back to the drawer. |
| 15:45.54 | elguero | if you are using a raw format, then there wouldn't be any headers, correct |
| 15:47.50 | elguero | a configurable minimum length would be nice as a new feature... but for now, we have to remove the 0 length audio files that are being created |
| 15:48.07 | elguero | thanks for your input, its appreciated :) |
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| 21:37.04 | pabelanger | Certified asterisk bugs need to be reported to Digium directly, right? |
| 21:37.30 | pabelanger | At least, that is how I remember talking about it |
| 21:37.57 | Qwell | I don't believe that's the case, no. |
| 21:40.03 | pabelanger | http://www1.digium.com/en/products/asterisk/certified-asterisk says to report them to Digium direct |
| 21:41.14 | Qwell | Where are you seeing that? |
| 21:42.25 | jsmith | Maybe in the link he just posted? |
| 21:42.28 | jsmith | ducks |
| 21:42.42 | pabelanger | Q: How would I submit an issue report for Certified Asterisk? |
| 21:43.06 | Qwell | Do you have an SLA? |
| 21:43.14 | pabelanger | no |
| 21:43.21 | pabelanger | somebody opened a ticket on the issue tracker |
| 21:43.26 | pabelanger | I redirected them to Digium |
| 21:43.26 | Qwell | Then it's not relevant. |
| 21:44.10 | putnopvut | Yeah, certified asterisk is just another version of Asterisk that gets bugs reported against it like any other. If you have an SLA with Digium, then you must use certified Asterisk as part of the SLA. |
| 21:44.42 | putnopvut | But if you don't have an SLA with Digium, then it gets put in the bugtracker just like any other bug. |
| 21:45.10 | malcolmd | and, as noted by the answer to that question on the web page, issues reported by non-sla customers do not trigger point releases of certified asterisk |
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