| 00:13.52 | gtjoseph | rmudgett: still there? got a sec? |
| 00:14.11 | rmudgett | gtjoseph: hes |
| 00:14.13 | rmudgett | yes |
| 00:14.31 | gtjoseph | INTERNAL_OBJ was declared inline originally. you sure you want me to remove it? |
| 00:15.23 | rmudgett | yes. It never really needed to be declared inline. |
| 00:15.33 | gtjoseph | gotcha, thanks. |
| 00:16.01 | rmudgett | gcc does a good job of inlining functions by itself. |
| 00:16.45 | gtjoseph | yep it does. just didn't know if there was an issue that needed it forced sometime in the past. |
| 00:18.49 | rmudgett | Making gcc inline all that code likely could cause more cache misses since it isn't a trivial function. |
| 00:19.13 | gtjoseph | true |
| 00:30.31 | gtjoseph | rmudgett: do you want is_ao2_object to just wrap INTERNAL_OBJ or just a quick check of the magic number? |
| 00:32.53 | rmudgett | gtjoseph: wrapping would be better. |
| 00:33.01 | gtjoseph | gotcha |
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| 06:06.15 | oej | Morning folks! |
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| 07:26.16 | michael_work | oej, \o |
| 07:27.17 | michael_work | oej, i read your tweet few weeks ago about you playing with DTMF. remind me what functionality you was adding/improving? |
| 07:40.21 | wdoekes | oej: I was taking a peek at the pinefrog rtcp code. the 11 branch does not seem up to date, while the 1.8 version is up to date. I have no idea how the auto-svn-merge stuff works. is that set correctly? does it need manual love? |
| 07:54.47 | oej | the 11 branch should be removed, yes. The 1.8 is running all the time. The teapot branch contains a lot of that stuff too. |
| 07:55.07 | oej | Isn't pinefrog-rtcp-1.8 up to date? It should be. We use it in production in a *lot* of sites. |
| 07:58.53 | wdoekes | 1.8 *is* up to date |
| 08:00.12 | wdoekes | my client wanted me to poke the asterisk team about call quality monitoring |
| 08:00.51 | oej | That branch writes CQR records for each call leg. |
| 08:00.59 | oej | And sends out information on manager. |
| 08:01.18 | oej | It's been around for five years and has been *very* useful. |
| 08:01.27 | oej | The problems are telephones that doesn't play RTCP. |
| 08:01.43 | wdoekes | what would you like to add/change before attempting a merge? |
| 08:01.58 | wdoekes | (into trunk) |
| 08:02.30 | michael_work | so we might see it next 1.8 releases? |
| 08:02.47 | michael_work | btw isn't 1.8 lifetime is getting to the end? |
| 08:03.00 | wdoekes | michael_work: no new features into release branches, so not into 1.8 |
| 08:03.04 | oej | No, we never add new functions to a release. You may see it in Asterisk 14 if the dev team wakes up. |
| 08:03.25 | oej | Yes, 1.8 lifetim is getting to the end and we're just about to migrate our sites to 1.8 |
| 08:03.42 | oej | It has gotten useful during the last half year and we have migrated a lot of old patches to it. |
| 08:03.44 | michael_work | i'm not sure i'll be alive when we'll get 14 :P |
| 08:04.04 | oej | teapot-1.8 contains a lot of patches that may be interesting for 1.8 users |
| 08:04.13 | oej | Looking for funding to get rtcp into 11 and trunk. |
| 08:04.23 | michael_work | i'm pushing migration to 11 so it would be easier to migrate to 13 :P |
| 08:04.42 | oej | The difference betwen 11 and 13 will be HUGE. |
| 08:04.46 | wdoekes | oej: it's just a matter of forward porting pinefrog? |
| 08:04.55 | michael_work | but less than 1.8 vs 13 |
| 08:05.01 | oej | If it took us many years to plan, test and start migration to 1.8 it will be very hard to get to 13. |
| 08:05.24 | oej | Not much. The difference between 1.8 and 11 is not that big. |
| 08:05.54 | michael_work | there are few usefull addons. some changes. at least for what we use |
| 08:06.01 | oej | Asterisk 12 introduced a lot of new code, that still needs a lot of testing to reach maturity. And the PJSIP channel still lacks functionality. |
| 08:06.20 | wdoekes | I suppose it isn't. but regarding CQR features, is anything missing in pinefrog that must be included? |
| 08:06.25 | michael_work | btw from what version pjsip gonna step in? |
| 08:06.43 | oej | I am not saying that Asterisk 12 isn't cool. It contains stuff we should have fixed many years ago and is a big step fowrard for Asterisk. It's a very important release. |
| 08:06.57 | oej | PJSIP was part of 12. |
| 08:07.10 | oej | But from my point of view, it's still a lot of work in progress. |
| 08:09.43 | wdoekes | oej: can I go ahead and assume that you say that the pinefrog features are complete enough to be merged (apart from it needing to be forward ported, obviously)? |
| 08:10.46 | wdoekes | or am I better off using a different branch of yours? (only focusing on rtcp and call quality) |
| 08:12.29 | oej | I would love some feedback on the code from someone like you, walter. |
| 08:12.45 | oej | But we use it in production, yes. It's part of the Lync certification I've been working with. |
| 08:13.10 | oej | Also used in call centers, some pretty large ones, since many years (on 1.4) |
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| 08:16.31 | wdoekes | ok, thanks for the info :) |
| 08:18.17 | oej | Feel free to contact me directly with any questions. I would love some support for that branch getting merged. Mjordan has been spending time with it so he is interested. |
| 08:18.36 | oej | Digium could do great things since they control both phones and asterisk, it surprises me that they don't. |
| 08:18.54 | oej | I would love to be in that position. |
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| 08:28.43 | michael_work | oej, the question i asked earlier about DTMF. i just wonder as i needed to set duration manually for inband. and i used dirty hack of taking parameter from channel's global variables. as i recall you mentioning something related to duration - wanted to ask, what exactly you was playing with |
| 08:30.27 | oej | I have rewritten a large part of the duration handling so we don't change the duration in transmit |
| 08:30.40 | oej | Incoming duration matches outbound much better. |
| 08:34.28 | michael_work | i think there should be on channel option or per device. as some devices do not recognize short dtmfs. e.g. we have few analog pancodes that do not recognize dtmf if it's not long enuf |
| 08:35.02 | oej | Wonder if qualify ever worked over TLS. Asterisk opens connection, sends OPTIONS, then immediately close the TLS/TCP connection. The other end have no connection to respond on. |
| 08:35.19 | michael_work | so what i basically did is added chan param for duration . and i check if it exists and it's valid and biger than minimum - so i use it |
| 08:35.39 | oej | The minimum duration could be settable that way. Right now I've made it into a global variable. |
| 08:35.47 | oej | It was hardcoded before. |
| 08:35.54 | michael_work | yes i know |
| 08:36.09 | oej | That should be fairly easy to code. |
| 08:36.15 | michael_work | but by gloal variable you mean per server (global) |
| 08:36.23 | michael_work | in my case it's per channel |
| 08:36.31 | oej | It's a setting in asterisk.conf, so yes, it's per pbx. |
| 08:37.19 | michael_work | i think it should be per channel. as when it fixes problems with some devices it may cause problems with others |
| 08:38.06 | oej | then we could also make it an per-device option in sip.conf |
| 08:38.25 | oej | hack on! |
| 08:41.02 | michael_work | :) |
| 08:41.24 | michael_work | i wanted to read a little bit more about DTMF over rfc and sip info first |
| 08:42.05 | michael_work | as i don't understand how duration works there. I only used my dirty patch for inband :) |
| 08:46.40 | oej | No, you don't need to learn anything about SIP Info and DTMF. That's is deprecated and should NOT be used At all. |
| 08:46.58 | oej | Over 2833/4733 RTP is quite easy |
| 08:47.21 | oej | We get a first packet, then one every 20 ms (based on the packetization of the audio) and after that three end packets |
| 08:47.30 | oej | Every packet updates the duration. |
| 08:47.56 | oej | My patch makes sure we send an AST_FRAME_DTMF for every RTP packet we receive over the bridge to update the duration. |
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| 09:55.18 | michael_work | if i want to send something for review |
| 09:55.48 | michael_work | do i need to use current branch or i can use certain version as well? |
| 11:23.27 | wdoekes | you should use the latest revision of the lowest/oldest branch that the change applies to |
| 11:23.42 | wdoekes | if it is a bugfix, it should be against 1.8 (if the bug exists there) |
| 11:23.54 | wdoekes | if it is a new feature, it should be against trunk |
| 11:24.12 | wdoekes | @ michael_work |
| 11:25.02 | michael_work | i can't add feature to 1.8 anymore? |
| 11:38.05 | file | new features are not permitted in 1.8 or 11, and only of a limited scope in 12 |
| 11:39.53 | michael_work | arrrrg :( |
| 11:40.04 | michael_work | it's not cool :( |
| 11:40.22 | michael_work | and if it's improvement and not really a new feature? |
| 11:40.35 | file | only bug fixes |
| 11:40.41 | michael_work | :_( |
| 11:41.00 | michael_work | https://issues.asterisk.org/jira/browse/ASTERISK-23823 - at least if someone want, he can grab it from jira |
| 12:02.08 | opticron | anyone feel like reviewing a SIPp integration test for REFER/Replaces transfers with a little 3PCC thrown in? https://reviewboard.asterisk.org/r/3560/ |
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| 15:43.36 | sruffell | seanbright: if you're around I wonder what you think about https://github.com/sruffell/dahdi-linux/commit/5e28debd349a2840 |
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| 16:58.40 | tm1000 | rmudgett: mjordan thanks for commit the RB |
| 16:59.12 | tm1000 | newtonr: speaking of which, I dont have commit access, which is fine. Just pointing that out for future reference. I let the cool kids do it, rmudgett mjordan or file |
| 16:59.33 | newtonr | tm1000, thanks for letting me know. i thought you did |
| 16:59.38 | file | being cool is no longer cool |
| 16:59.49 | tm1000 | newtonr: I dont commit enough for that |
| 19:09.00 | gtjoseph | rmudgett: The astobj2 split is ready for review again if you have a chance. |
| 19:09.58 | rmudgett | gtjoseph: Was just starting to look at it now |
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| 21:50.39 | gtjoseph | rmudgett: i grouped some things in AST_DEVMODE blocks that were previously separated, would you rather have structures and functions individually ifdef'd even if they were right next to each other? |
| 21:52.27 | gtjoseph | ...and i'm just realizing I have #if collapsing turned on which is why I've been missing some of things that I cut and pasted. :( |
| 21:52.50 | rmudgett | gtjoseph: functions should be individually ifdeffed since that is usually what you need to know if it is active or not. Declarations since they are usually not very large don't need to be individually ifdeffed. |
| 21:53.23 | gtjoseph | gotcha |
| 21:54.17 | rmudgett | I've been comparing the split files with the original. |
| 21:54.36 | rmudgett | It makes the changes stand out. |
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| 22:39.12 | gtjoseph | rmudgett: if your eyes aren't already bleeding (mine are), I just pushed another patch up. :) |
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| 23:43.56 | gtjoseph | rmudgett: pushed again. |
| 23:45.25 | rmudgett | gtjoseph: Heh. I'm trying to compile it under different options. You need to include paths.h in astob2 when REF_DEBUG is enabled. |
| 23:45.51 | gtjoseph | ah |
| 23:46.02 | rmudgett | Ironically refcounter won't compile if REF_DEBUG is defined. |
| 23:46.26 | gtjoseph | what does it complain about? |
| 23:46.52 | rmudgett | ast_config_AST_LOG_DIR not being declared. It doesn't link. |
| 23:48.18 | gtjoseph | huh, in the compile of the copied astobj2.c has the same problem. ?? |
| 23:48.50 | rmudgett | It compiles it just doesn't link |
| 23:49.28 | gtjoseph | for me refcounter is ok, but astobj2.c isn't which doesn't make sense since refcounter is dependent on astobj2.o. maybe i need to do a distclean. |
| 23:49.47 | rmudgett | I think this is a new problem since REF_DEBUG was added to menuselect recently. |
| 23:50.16 | gtjoseph | let me try on a stock 12 |
| 23:50.51 | rmudgett | I think stock is going to have this problem |
| 23:53.12 | gtjoseph | yep it does |
| 23:55.30 | rmudgett | It's going to be on all current branches too. |
| 23:56.00 | gtjoseph | :( |
| 23:56.16 | gtjoseph | want me to track it down as part of this patch? |
| 23:57.15 | rmudgett | no since it would need to be done for v1.8, 11, and 12 |
| 23:57.38 | gtjoseph | ah, i dodn't know ref_debug was backported |
| 23:58.17 | rmudgett | mjordan put it on all branches because it is easier to say enable it and give post the log. |