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| 06:09.43 | jmls | morning all! |
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| 06:15.25 | schmidts | good morning |
| 06:15.33 | irroot | lo there |
| 06:19.05 | schmidts | irroot how are you today? and how is the weather at yours? |
| 06:20.17 | irroot | Mist broken clouds 7c dew point 6c humidity 93% wind 7.2m/s 1025hPa <- winter's day |
| 06:20.32 | irroot | all good otherwise have a good day |
| 06:21.00 | schmidts | its nearly the same over here ;) only 14?C ,rain and heavy wind but ITS SUMMER HERE!!! :( |
| 06:22.09 | irroot | it will get to 10-15 latter i guess its still early sun been up < 1hr i cant handle the cold much :P |
| 06:22.32 | irroot | summer its 25-35 most days |
| 06:22.48 | schmidts | normally its the same here ;) |
| 06:23.13 | irroot | ah well then its good |
| 06:23.27 | schmidts | but i am afraid of asking oej how is it up in the north, cause i guess its even warmer there now :D |
| 06:23.54 | oej | he he... |
| 06:23.57 | oej | Not today |
| 06:24.16 | schmidts | :D |
| 06:24.59 | irroot | maybe break out a bottle of sherry or something for the cold .... |
| 06:25.41 | schmidts | bad idea whlie i am at office ;) |
| 06:26.50 | irroot | mmm maybe not i might be more plesant but less productive |
| 06:27.32 | irroot | ole ever see the blink soft phone ?? if so any comments as to if its a good phone ie RFC complient |
| 06:27.38 | schmidts | :D something like this |
| 06:36.51 | irroot | the mobile networks hate me now ... have too RICA all sim cards now Registration and Interception of Comunications Act .... basically a massive .gov database of home addresss ID and sim/cell numbers |
| 06:37.39 | irroot | im not about to do this |
| 06:45.07 | oej | I think the Blink softphone is great. |
| 06:45.13 | irroot | awesome thx |
| 06:45.32 | oej | Btw, no biggie but I'm "Olle" not "Ole". "Ole" is a Danish name ;-) |
| 06:45.53 | irroot | sorry ... |
| 06:46.05 | oej | Where are you based, irroot? |
| 06:46.14 | irroot | zuid africa |
| 06:46.23 | irroot | johannesburg |
| 06:47.15 | schmidts | Oej isnt Ole more like spanish? |
| 06:50.38 | irroot | anyone seen any calls hanging in pbx with no locks after a pickup ?? |
| 06:54.19 | sgimeno | Olé is Spanish ;) |
| 06:56.05 | irroot | i so failed german / afrikaans at school could never work out the accent characters way too confusing |
| 06:57.06 | oej | schmidts: THat's olé ! |
| 06:57.49 | oej | Have really weird crashes that I suspect is related to voicemail notification |
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| 07:10.38 | irroot | me wonders who gave kpfleming some of the good stuff to smoke he mentioned you can set / unset faxdetect from func_channel ?? am i missing somethin |
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| 08:34.34 | schmidts | irroot you maybe missed the stuff he got ;) |
| 08:35.00 | irroot | appears so |
| 08:35.31 | irroot | im looking for a better way of detecting faxes dealing with faxdetect in the dialplan |
| 08:35.56 | irroot | he said i can disable it with CHANEL(faxdetect) i dont see the code scotty |
| 08:36.19 | schmidts | maybe he meant you should write this function ;) |
| 08:36.34 | irroot | mmm i think thats what he meant now |
| 08:36.53 | irroot | i may do it i did it for OOH323 in me branch |
| 08:37.18 | irroot | ill port it to func_channel and use queryoptions/setoptions ?? |
| 08:39.41 | schmidts | irroot you mean ast_channel_setoption? IMHO why not |
| 08:40.39 | schmidts | but give me a second i have to look for something |
| 08:40.47 | irroot | np |
| 08:42.15 | schmidts | i havent follow your code very deeply but will you use a framehook for the faxdetection or something else? |
| 08:43.05 | schmidts | cause if its a framehook everything is allready their so it should be easy adding the faxdetect option to func_channel.c in func_channel_read and then just add the framehook to the channel |
| 08:43.27 | irroot | the framehook is not needed for my WaitFax its blocking the channel faxdetect is in the background |
| 08:43.41 | irroot | i wanted a blocking function for this not background |
| 08:44.23 | irroot | the problem with the background faxdetect is it can rip a call from a fax and route it back and gets a engaged |
| 08:44.53 | irroot | i need to turn faxdetect off on a channel if i know its going to a fax terminal |
| 08:45.23 | irroot | call comes in on E1 it has faxdetect for things like fax to mail group fax machine routing .... |
| 08:47.18 | irroot | if there is a FXS / ATA on the network that is dedicated the faxdetect re routes the call |
| 08:53.10 | schmidts | irroot ok i just asked cause you will also have to add a var to the ast_chan structure if you want to set the faxdetect option in func_channel ;) |
| 08:56.40 | irroot | yip |
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| 09:01.59 | irroot | on conf call with loyds bank implementing a asterisk solution in Johannesburg And Cape Town |
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| 09:15.46 | irroot | schmidts ill look into it think its a nice to have and if channel does not support it need to be handled |
| 09:17.14 | irroot | AST_OPTION_FAXDETECT |
| 09:20.17 | irroot | ok will do this what channels need it 1'st DAHDI/SIP/mISDN/OOH323/IAX .... |
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| 11:27.12 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- http://xkcd.com/303/ |
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| 11:41.13 | irroot | kpflemming morning ... |
| 11:41.37 | irroot | i looked at the source and there is no CHANNEL(faxdetect) |
| 11:42.03 | irroot | ill be adding it to work with DAHDI / mISDN / SIP / IAX / OOH323 initially |
| 11:42.18 | irroot | using _setoption / _queryoption |
| 11:46.55 | kpfleming | well, you could do it that way, or just implement it in the CHANNEL function callbacks in the channel driver |
| 11:46.58 | kpfleming | but setoption() is ok too |
| 11:48.25 | irroot | cool thx |
| 11:57.32 | irroot | needs too make plans for +2 to go too astricon |
| 12:08.23 | kpfleming | nice |
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| 14:38.26 | cemeh4 | hi all! |
| 14:38.58 | cemeh4 | i'm implementing rtmp channel and sip -> rtmp application |
| 14:39.46 | cemeh4 | can somebody help me with asterisk's bridging interface? |
| 14:40.10 | cemeh4 | how can'i properly connect 2 or more channels? |
| 14:40.42 | cemeh4 | is there a way to automatic translate data between various formats? |
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| 14:54.53 | jsmith | cemeh4: I'm assuming you're aware of https://issues.asterisk.org/view.php?id=15484 |
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| 14:57.09 | cemeh4 | i have taked this code as starting point, but currently almost everything is rewrited |
| 14:57.57 | cemeh4 | ... sorry for my english :-( |
| 15:01.08 | cemeh4 | i removed ffmpeg at all, i'm using only librtmp from rtmpdump |
| 15:01.29 | cemeh4 | channel is working now |
| 15:01.59 | cemeh4 | anly audio supported |
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| 15:51.06 | Kobaz | okay so |
| 15:51.10 | Kobaz | i want to go and commit something now |
| 15:51.20 | Kobaz | is there a commit template in jira? |
| 15:51.27 | kpfleming | there is not yet |
| 15:51.36 | Kobaz | i should copy one from mantis i guess |
| 15:51.44 | kpfleming | you can, or you can just copy one from a recent commit |
| 15:51.47 | Kobaz | ASTERISK-17911 |
| 15:51.48 | Kobaz | k |
| 15:51.49 | CaptainPants | [ASTERISK-17911] [Status: Open] [patch] Ability to dynamically add filters to an ami session - https://issues.asterisk.org/jira/browse/ASTERISK-17911 |
| 15:52.07 | Kobaz | i updated my reviewboard diff per the comments, i'm pretty sure it's all good |
| 15:52.14 | irroot | cheers folks see you on flip side on way home |
| 15:52.49 | Kobaz | oh |
| 15:52.56 | Kobaz | and where do i get my dev permissions on jira? |
| 15:54.05 | Kobaz | SWP-3519 |
| 16:02.03 | kpfleming | you can't see or work with SWP issues |
| 16:02.06 | kpfleming | what do you need to do? |
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| 16:14.31 | Kobaz | delete a patch |
| 16:14.39 | Kobaz | the old patch i put in |
| 16:15.01 | Kobaz | i wasn't specifically interested in SWP, noticed it was marked a relation, when i didn't have access i figured it was internal |
| 16:15.10 | Kobaz | and i want to change the asterisk version from 1.4 |
| 16:15.42 | Kobaz | the patch is based on trunk, and hopefully will go into 1.10 |
| 16:17.13 | kpfleming | ok, so you need bug marshal abilities then |
| 16:17.23 | kpfleming | what is your JIRA username? |
| 16:17.24 | Kobaz | sure... i think i had that in mantis |
| 16:17.25 | Kobaz | kobaz |
| 16:18.28 | kpfleming | ok, you've been added to the appopriate group... but due to caching in the JIRA backend, it could take an hour or so to be effective |
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| 16:18.46 | Kobaz | ah, k, cool |
| 16:20.29 | Kobaz | should i add to CHANGES ? |
| 16:20.39 | kpfleming | if you are adding a feature, yes |
| 16:20.42 | Kobaz | okay |
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| 16:46.42 | Kobaz | what about post-commit... Should I manually close the issue? |
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| 16:57.31 | irroot | kobaz copy the commit message into the issue and close it mannually .... |
| 16:57.55 | irroot | wonders when russellb will complete the template script :P |
| 16:59.41 | Kobaz | k |
| 17:05.20 | Kobaz | done and done |
| 17:05.25 | Kobaz | four more patches to go |
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| 18:37.32 | Kobaz | kpfleming: i don't think my permissions kicked in |
| 18:38.32 | kpfleming | hmm |
| 18:38.35 | kpfleming | have you logged out and back in? |
| 18:39.00 | kpfleming | i don't quite understand how the user object caching in JIRA works, unfortunately |
| 18:39.35 | kpfleming | but if i look in JIRA, it does show the proper group memberships for you |
| 18:50.59 | Kobaz | k |
| 18:51.06 | Kobaz | i did try logout and in, a bit ago |
| 18:51.08 | Kobaz | i can try again |
| 18:51.27 | kpfleming | ok. exactly what are you trying to do that you are denied? |
| 18:52.49 | Kobaz | i was gonna delete the old patch for one of my issues |
| 18:52.53 | Kobaz | i dont see any options for it |
| 18:53.04 | kpfleming | that might not be allowed except for admins |
| 18:53.08 | Kobaz | ah |
| 18:53.12 | kpfleming | i don't know to be honest |
| 18:53.25 | Kobaz | i generally clean up my tickets so that the patch in there matches the reviewboard and etc |
| 18:53.27 | kpfleming | leifmadsen: do you know what's allowed? |
| 18:53.57 | Kobaz | unless there are differeing approaches, then i would leave those in |
| 18:54.07 | leifmadsen | kpfleming: I'm not sure what is allowed for people who aren't admins, but I know we didn't allow people to delete patches who weren't admins in mantis |
| 18:54.09 | leifmadsen | so that doesn't surprise me |
| 18:54.16 | leifmadsen | what is the issue number? |
| 18:54.28 | Kobaz | i just closed it, sec |
| 18:54.51 | Kobaz | ASTERISK-17911 |
| 18:54.52 | CaptainPants | [ASTERISK-17911] [Status: Closed] [patch] Ability to dynamically add filters to an ami session - https://issues.asterisk.org/jira/browse/ASTERISK-17911 |
| 18:55.27 | Kobaz | it's not a huge deal, but it makes it easier to work with if you need to reference the issue in the future |
| 18:55.33 | Kobaz | like, which is the official patch for the issue |
| 18:55.36 | leifmadsen | were you in "Manage Attachments" ? |
| 18:55.42 | Kobaz | hmm |
| 18:55.48 | leifmadsen | I'm just curious if you were in the right area at all :) |
| 18:56.09 | leifmadsen | beside Attachments on the far right, beside the + there is a down arrow, do you see that? |
| 18:56.14 | Kobaz | don't have a 'manage attachments' |
| 18:56.18 | Kobaz | yeah the + |
| 18:56.30 | leifmadsen | bottom of the list, you don't have Manage Attachments? |
| 18:56.32 | Kobaz | i can hit the + and add a new file |
| 18:56.33 | Kobaz | nope |
| 18:56.38 | leifmadsen | Download All? |
| 18:56.50 | Kobaz | nope |
| 18:56.57 | leifmadsen | odd that wouldn't be enabled by default |
| 18:57.00 | Kobaz | oooooooh |
| 18:57.06 | Kobaz | the \/ RROW |
| 18:57.09 | Kobaz | arrow |
| 18:57.10 | Kobaz | rather |
| 18:57.12 | leifmadsen | yes, the down arrow :) |
| 18:57.16 | leifmadsen | as I stated :) |
| 18:57.23 | Kobaz | oh right, somehow i missed that |
| 18:57.32 | leifmadsen | my sentence was potentially unclear |
| 18:57.37 | leifmadsen | documenting is hard |
| 18:57.47 | Kobaz | ah okay perfect |
| 18:57.47 | leifmadsen | anyways, in the drop down menu, what options do you get? |
| 18:57.52 | Kobaz | yeah i see all those |
| 18:57.57 | Kobaz | manage attachments |
| 18:58.00 | Kobaz | download all |
| 18:58.02 | leifmadsen | ok, so under Manage Attachments you can delete from there |
| 18:58.10 | Kobaz | yeap, i saw |
| 18:58.19 | Kobaz | okay, axed the old one |
| 18:58.19 | leifmadsen | you should see a trashbin icon |
| 18:58.19 | leifmadsen | ok perfect |
| 18:58.47 | Kobaz | okay good, i see i can link issues and stuff |
| 18:59.22 | leifmadsen | :) |
| 18:59.26 | leifmadsen | \o/ |
| 18:59.28 | leifmadsen | my work here is done |
| 19:00.16 | Kobaz | hehe |
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| 19:16.29 | jsmith | Hey oej! |
| 19:20.19 | oej | Hello Jared! Where on the globe are you today? |
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| 19:33.24 | jsmith | oej: Back at home in Virginia, believe it or not :-) |
| 20:06.09 | oej | jsmith: ...and I'm spending one week back home too... |
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| 21:21.43 | cemeh4 | hi all! |
| 21:22.46 | cemeh4 | i'm implementing rtmp channel and sip -> rtmp application. can somebody help me with asterisk's bridging interface? |
| 21:23.12 | kpfleming | isn't there already an RTMP channel driver out there from i6net? |
| 21:23.55 | cemeh4 | 1) how can'i properly connect 2 or more channels? 2) is there a way to automatic translate data between various formats? 3) how can i properly monitor bridged channel's state (AST_STATE_DOWN, AST_STATE_UP etc.)? |
| 21:25.24 | kpfleming | you really want to use ConfBridge to connect your channels together, i expect |
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| 21:25.37 | kpfleming | you don't want to build an entire channel bridging loop yourself... it's rather complicated |
| 21:27.02 | cemeh4 | i need to implement custom conference algorithm |
| 21:27.30 | kpfleming | then you need to build a new bridge module, like bridge_softmix |
| 21:27.32 | cemeh4 | one sip channel to multiple RTMP endpoints |
| 21:28.39 | kpfleming | asterisk can't do that... it can't handle multiple media streams on a single channel |
| 21:28.40 | cemeh4 | rtmp endpoints dinamically connecting/disconnecting while sip channel up all application's lifetime |
| 21:28.58 | kpfleming | you are basically talking about an entire asterisk core (for media handling) inside your module |
| 21:29.51 | cemeh4 | basically it is working now, but there is some issues |
| 21:30.09 | cemeh4 | 1) i cant handle when sip channel disconnects |
| 21:32.04 | cemeh4 | 2) i cant properly impart/depart new channels to the ast_bridge, i can't understand how it working internally and now, sometimes, i have segfaults on ast_bridged_channel calls inside my bdrige |
| 21:32.09 | kpfleming | umm.. if you have one sip channel handling multiple RTMP channels, how is it even possible to keep things alive if the sip channel gets disconnected? |
| 21:32.32 | kpfleming | ast_bridge doesn't handle more than 2 channels |
| 21:32.48 | cemeh4 | i want properly shutdown internal structures on sip disconnect and dispose my bridge |
| 21:32.49 | kpfleming | what you are trying to do sounds very, very confusing |
| 21:34.01 | cemeh4 | ast_bridge has flag AST_BRIDGE_CAPABILITY_1TO1MIX and AST_BRIDGE_CAPABILITY_MULTIPLE (or something like that) |
| 21:34.14 | kpfleming | oh, i was thinking of something else |
| 21:34.28 | kpfleming | but yeah... it will handle only multiple real asterisk channels, not multiple media streams for one channel |
| 21:35.17 | cemeh4 | i have created RTMP channel (based on asterisk branch with basic rtmp support) |
| 21:35.40 | cemeh4 | i have rewrited it |
| 21:35.41 | kpfleming | then send that channel into the dialplan and run ConfBridge to add it to a conference with the other channels |
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| 21:36.47 | cemeh4 | currently i'm dinamically creating and destroying RTMP channels in my conerence app |
| 21:37.10 | cemeh4 | depends on what rtmp stream becomes active |
| 21:37.37 | cemeh4 | i'm getting this info from our external server |
| 21:37.51 | kpfleming | what you are doing is far too complex to try to help you with here, unfortunately |
| 21:39.47 | cemeh4 | actually i have some basic questions on what i can't find solutions from existing code (i'm investigated only app/chan/codecs folders, asterisk's internals too complex for me) |
| 21:40.12 | kpfleming | well, handling channel threads is a very complex thing |
| 21:40.34 | kpfleming | you can't just create channels and dump them into a bridge.... they must have threads to own them and process their audio |
| 21:40.59 | kpfleming | those threads are also responsible for cleaning up when the channel needs to be destroyed and other stuff like that |
| 21:44.02 | cemeh4 | actually channel is working now, i can, for example, write it in sip.conf as endpoint for incoming sip calls and it working normally (there is issue with decoding speex data packets with 2 speex frame in it, but if i set on sender side data length 20ms it is working normally) |
| 21:44.49 | kpfleming | if you use the asterisk channel mechanisms properly, you don't need to deal with any of that. all of that will be handled properly for you. |
| 21:45.21 | cemeh4 | that question :-) |
| 21:46.07 | kpfleming | every media stream needs to be its own ast_channel, probably with an ast_pbx running it, that then runs an instance of a conference application to join whichever conference it is supposed to join |
| 21:46.39 | kpfleming | note that most of this is in relation to SVN trunk (soon to be asterisk 1.10)... the media and bridge code in asterisk 1.8 is very different, and not as flexible or powerful |
| 21:47.00 | *** join/#asterisk-dev mpe_ (~mpe@212.45.120.202) |
| 21:49.15 | Corydon76-home | Is Russell out of office today? |
| 21:49.27 | cemeh4 | i'm creating ast_channel and starting ast_pbx |
| 21:49.33 | kpfleming | Corydon76-home: yes |
| 21:49.43 | kpfleming | cemeh4: for every channel, both SIP and RTMP? |
| 21:50.36 | Corydon76-home | kpfleming: Are you able and comfortable with giving me commit permissions to the menuselect repo? |
| 21:50.56 | kpfleming | i believe so... anything you do there is reversible :-) |
| 21:51.13 | Corydon76-home | Heh. Please? |
| 21:51.21 | cemeh4 | sip - is caller started by dialplan and injected to my application as argument of ast_exec |
| 21:51.56 | kpfleming | Corydon76-home: unless i've done it wrong, you should have it now |
| 21:52.00 | Corydon76-home | Danke |
| 21:52.01 | cemeh4 | i'm managing RTMP channels only |
| 21:53.05 | cemeh4 | another problem is codecs |
| 21:53.51 | cemeh4 | when sip call is sstarted from phone which not support speex |
| 21:54.23 | cemeh4 | is there a way to automatic conversion between formats |
| 21:54.25 | cemeh4 | ? |
| 21:54.32 | kpfleming | what does 'automatic' mean? |
| 21:54.45 | kpfleming | at the code level, no, you have to tell the channel what format you want to read/write audio in |
| 21:55.12 | kpfleming | but if you are mixing audio, you must be already doing that |
| 21:56.35 | cemeh4 | small example: sip call started using g722. on some event i'm starting rtmp channel with speex codec and trying to impart both to custom bridge |
| 21:57.13 | kpfleming | then your 'custom bridge' needs to use the proper API calls to set the channel read and write formats appropriately |
| 21:57.22 | kpfleming | look in channel.h and format.h |
| 21:57.23 | cemeh4 | and in debug i see something line: frame speex but read format is g722 |
| 21:58.02 | cemeh4 | it means that there is a way? |
| 21:58.04 | cemeh4 | ok |
| 21:58.19 | cemeh4 | another issue: |
| 21:58.22 | kpfleming | of course there is a way... did you think that asterisk could only connect channels of the same format? |
| 21:59.09 | cemeh4 | i'm spent 2 days to deal with that |
| 22:00.16 | kpfleming | and you didn't find ast_set_read_format() and ast_set_write_format()? |
| 22:00.17 | cemeh4 | another issue: when physically hangup occurs |
| 22:00.44 | cemeh4 | :-) great thx |
| 22:00.55 | cemeh4 | no i did not... |
| 22:01.48 | cemeh4 | in debug i see something like: sip recieved bye event lalala setting device state to 0 |
| 22:02.01 | kpfleming | it's documented in channel.h. i'm surprised you didn't come across it. |
| 22:03.25 | cemeh4 | inside application's loop i'm cheking sip channel's member _state |
| 22:04.12 | cemeh4 | and it's state is AST_STATE_UP even after bye event |
| 22:04.17 | kpfleming | you should be calling ast_check_hangup() or something like that |
| 22:04.35 | kpfleming | structure variables with underscore prefixes are not meant to be used in your code, they are internal |
| 22:05.20 | cemeh4 | very thx |
| 22:05.42 | cemeh4 | one more question about brige |
| 22:05.49 | kpfleming | you should go look at an existing simple application to see how it handles knowing when the channel has been hung up. app_echo or app_playback, for example. |
| 22:06.37 | cemeh4 | i have looking there |
| 22:07.30 | cemeh4 | ... maybe i miss something |
| 22:07.49 | kpfleming | in app_echo, for example, ast_waitfor() will return -1 when the channel has been hung up |
| 22:08.39 | cemeh4 | in what cases channel must be locked? |
| 22:08.46 | kpfleming | which is also documented in channel.h, although it says 'failure'... but clearly the channel being hung counts as failure for it to be ready to read/write |
| 22:08.54 | kpfleming | that's a very large question |
| 22:09.08 | cemeh4 | ok |
| 22:09.13 | Corydon76-home | The channel must always be locked when a change, explicit or implicit, will be made to the attributes thereof |
| 22:09.14 | kpfleming | most of the API calls are documented to tell you whether you must hold the channel lock or not when calling them |
| 22:13.20 | kpfleming | Corydon76-home: can't menuselect automatically determine whether a <use> dependency is a module or an external? |
| 22:13.21 | cemeh4 | in application i'm prepearing all data inapplication's thread then starting new thread(to allow main application thread continue handling messages from our server) and there calling ast_call(myrtmp) and in case of cuccess ast_bridge_impart (here is helper thread end), but when i'm trying to hangup rtmp asterisk warning me about: trying to hangup in thread NNN, but fd is locked by thread XXX? |
| 22:13.45 | Corydon76-home | kpfleming: menuselect did not have visibility to that, no |
| 22:13.55 | kpfleming | if you think the asterisk core is too complex for you, then you're working on pretty complex stuff now |
| 22:14.22 | Corydon76-home | kpfleming: unless you mean by the underscore in the name, which is a tenuous grasp and could easily fail |
| 22:14.26 | kpfleming | Corydon76-home: i think i put in code to scan the list already to determine that... when you are in menuselect, you can see whether a depedency is marked with (M) or (E) |
| 22:14.31 | cemeh4 | i did not see realisation files |
| 22:15.11 | cemeh4 | of course i have investigated header files |
| 22:15.14 | Corydon76-home | kpfleming: that's only for dependencies, not uses |
| 22:15.34 | Corydon76-home | Depends on: dahdi(E), tonezone(E) Can use: res_smdi, pri, ss7, openr2 |
| 22:16.15 | cemeh4 | i'm dealing with asterisk's api about one month |
| 22:16.58 | cemeh4 | i do not know all tricks with it, that is why i'm here |
| 22:19.29 | Corydon76-home | kpfleming: Ah, looks like it was there, but not used |
| 22:21.12 | cemeh4 | about bridge: what happend on bridhed channel hangup? do i need depart channel after hangup? |
| 22:22.10 | Corydon76-home | kpfleming: it's kind of a chicken and egg problem, because the question is whether we create a 'use' or a 'dep' struct at that point in the logic... and the 'member' member of the struct isn't populated until later |
| 22:23.03 | kpfleming | ahh |
| 22:23.05 | kpfleming | unfortunate |
| 22:23.40 | Corydon76-home | Something that could be done is to unify those structures and use a flag value to differentiate them |
| 22:23.56 | Corydon76-home | but I went with the smallest change, instead |
| 22:51.03 | kpfleming | Corydon76-home: sure... my only concern with that solution is introduces an opportunity for manual mistakes... it'd be nice to be able to avoid them, but not at the cost of having to do serious work on menuselect itself |