| 00:42.59 | russellb | leifmadsen: yes! |
| 00:51.42 | *** join/#asterisk-dev quintana (~sylvain@modemcable136.140-58-74.mc.videotron.ca) |
| 00:58.59 | quintana | hello, the ipv6 pjsip test in testsuite seems broken because the bind=[::1] is not accepted |
| 00:59.54 | quintana | [Oct 28 00:47:32] ERROR[1275] config_options.c: Error parsing bind=[::1] at line 7 of |
| 00:59.54 | quintana | [Oct 28 00:47:32] ERROR[1275] res_sorcery_config.c: Could not create an object of type 'transport' with id 'local-transport-tcp6' from configuration file 'pjsip.conf' |
| 01:00.21 | quintana | ./runtests.py -t tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_user |
| 01:02.19 | quintana | i used the trunk version of testsuite |
| 01:05.52 | file | is your pjproject built with IPv6 support? |
| 01:10.16 | quintana | hum that a good question, i check ! |
| 01:11.37 | quintana | i need a special configure to build with ipv6 support ? |
| 01:11.37 | snuff-work | putty |
| 01:13.25 | quintana | On Linux/Unix/MacOS X with GNU development systems, IPv6 features are detected by the ./configure script. |
| 01:13.33 | file | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject there's a part in there |
| 01:13.42 | file | "IPv6 support in pjproject is, by default, disabled. To enable it, add CFLAGS='-DPJ_HAS_IPV6=1' to your ./configure command." |
| 01:14.04 | quintana | ohhh thx :) |
| 01:14.09 | quintana | i missed that |
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| 01:31.28 | quintana | <?xml version="1.0" encoding="utf-8"?> |
| 01:31.29 | quintana | <testsuite errors="0" failures="0" name="AsteriskTestSuite" tests="1" time="8.42"> |
| 01:31.29 | quintana | <PROTECTED> |
| 01:31.29 | quintana | </testsuite> |
| 01:31.36 | quintana | thx file :) |
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| 10:33.01 | LooserOuting | hi. I was going to create ticket on jira about directrtpsetup in chan_sip.c. asterisk will add codecs to SDP if the called peer has more allowed codecs then offerred by caller. but this is wrong since asterisk cannot transcode in this case, right ? what do you think ? |
| 11:30.28 | LooserOuting | and has anybody thought of it storing the SDP in ast_channnel so if a channel suport SDP can make use of it. doing this is should be possible to use the saved SDP instead of creating a new one. (of course SDP could be altered so asterisk stays in the media path to do some kind of bridging). channel A is offering SDP to channel B. if channel B uses this option it has to inform channel A. |
| 11:37.45 | Stefan27 | how come the looping call to ast_sched_del in function __sip_ack in chan_sip.c doesn't use the AST_SCHED_DEL (which almost everything else uses) ? |
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| 12:54.03 | opticron | there's probably some locking or extra ref magic that has to happen in the middle of the DEL macro |
| 12:55.21 | opticron | yeah, the sip pvt has to be unlocked and relocked before and after the sleep |
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| 13:10.42 | Stefan27 | yeah but why doesn't it have that max-loop-iterations check |
| 13:14.07 | Stefan27 | i.e. why couldn't (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) stay true forever |
| 13:15.15 | Stefan27 | There's a long chunk of comments before the section i just don't quite get it :) |
| 13:20.17 | Stefan27 | because AST_SCHED_DEL does have that extra safe-check with count: while (id > -1 && (_sched_res = ast_sched_del(sched, id)) && ++_count < 10) ... |
| 13:21.49 | opticron | while it's possible that it could loop forever, that would likely mean there's already a deadlock anyway due to the retransmission requiring the pvt lock which would force the loop in question to wait forever for the pvt lock (in which case it really wouldn't be looping forever) |
| 13:22.46 | opticron | it might be good to have a loop count limiter in there anyway |
| 13:33.33 | Corydon76 | The scheduler desperately wants a rewrite |
| 13:34.50 | Corydon76 | The looping in there is basically because the API doesn't support the whole shebang being canceled |
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| 13:49.30 | Stefan27 | my asterisk did seem to get stuck there but i couldn't really ticket or debug the problem because when i had 'debug compiler flags on and logs enabled' my system never reached the point of failure because system was too slow. (around 50000 calls is usually needed and i have weak hardware). im just trying to get an overall-grip so i know where to collect debug info from. will return when |
| 13:49.30 | Stefan27 | i have more info |
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| 14:01.11 | opticron | Corydon76, it's on my personal todo list if nothing else |
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| 15:24.02 | mjordan | rmudgett: on ASTERISK-24437, which off nominal path is missing a call to ast_bridge_features_destroy? |
| 15:25.34 | mjordan | ah, neevermind. If there is a PBX and features were provided, then we don't destroy it |
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| 15:30.59 | JunK-Y | do we know how many people attend astricon this year? |
| 15:31.47 | jsmith | gtjoseph: New pjproject RPM being built in Koji for you to try: http://koji.fedoraproject.org/koji/taskinfo?taskID=7962692 |
| 15:32.01 | jsmith | gtjoseph: (and sorry it's taken so long -- I've been buried with travel the past week) |
| 15:32.21 | gtjoseph | jsmith: no worries! I'll give it a try when it's done. |
| 15:32.22 | leifmadsen | jsmith: huzzuh! |
| 15:32.36 | leifmadsen | now I can build my own RPMs against that spec file :) |
| 15:34.58 | jsmith | leifmadsen, gtjoseph: Also, http://copr.fedoraproject.org/coprs/jcollie/asterisk13/ |
| 15:35.08 | jsmith | Feedback welcome :-) |
| 15:35.13 | leifmadsen | very nice, I think I used that as a base |
| 15:35.26 | leifmadsen | then promptly ripped out all the modules that weren't named app_stasis |
| 15:35.27 | leifmadsen | :D |
| 15:38.47 | Marquis42 | :D |
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| 16:12.13 | mjordan | JunK-Y: Not sure yet |
| 16:16.49 | leifmadsen | mjordan: can you tell me more about removing the dialplan? |
| 16:18.50 | gtjoseph | leifmadsen: that's just mean. :) |
| 16:19.19 | leifmadsen | at least I waited till Tuesday and not on Monday |
| 16:19.30 | leifmadsen | I've used this joke 3 times now, so I will put it to bed :D |
| 16:19.38 | gtjoseph | ok, that's true |
| 16:23.50 | mjordan | heh. I look forward to my e-mails to the -dev list about AstriDevCon - which have to happen since we have a lot of good stuff to discuss |
| 16:23.56 | mjordan | Hopefully he'll get bored and wander away. |
| 16:24.11 | Marquis42 | One hopes |
| 16:25.59 | seanbright | that's hopeful |
| 16:26.11 | seanbright | just unsubscribe him from the list |
| 16:30.08 | file | little does everyone know I took this week off to remove the dialplan |
| 16:30.57 | seanbright | file: NICE |
| 16:33.12 | gtjoseph | let me know when you've got that extensions.conf to extensions.js converter done. |
| 16:34.00 | mjordan | rmudgett: does the patch I just threw onto ASTERISK-24437 look right/ |
| 16:36.25 | rmudgett | mjordan: You don't need to do a NULL check before calling ast_bridge_features_destroy(). It has its own NULL check. :) |
| 16:36.50 | rmudgett | Other than that yes it is ok. |
| 16:37.31 | mjordan | cool, I'll just go ahead and fix that. Working through the Triage list and picking off any easy low hanging fruit. |
| 16:37.47 | mjordan | things tend to pile up when the community is at AstriCon :-) |
| 16:44.11 | file | seanbright, I'm actually on the couch, under a blanket, drinking water and eating pasta salad, watching Ocean's 12, after taking suitable drugs to combat going from Vegas to my local environment! GOT'CHA! |
| 16:44.27 | seanbright | file: you totally fooled me! |
| 16:45.22 | file | seanbright, how's TTS? |
| 16:46.02 | dms1 | ya . . how is TTS? |
| 16:46.07 | dms1 | ;) |
| 17:08.16 | seanbright | in progress, but slowly |
| 17:08.31 | seanbright | dms1: i can move it to a team/group branch if you want to take a wack at it |
| 17:08.40 | seanbright | actually, let me do that anyway. |
| 17:09.19 | seanbright | dms1: took a look at app_swift, there are a few memory leaks i've found. i have a fork of it that i am working on. |
| 17:12.55 | seanbright | dms1: http://svnview.digium.com/svn/asterisk/team/group/generic-tts/ |
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| 17:18.08 | dms1 | sweet |
| 17:18.22 | seanbright | its a whole lotta nothing right now |
| 17:18.35 | seanbright | and i'm basically ripping off the generic speech API |
| 17:18.39 | dms1 | nice |
| 17:18.42 | dms1 | hehe |
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| 17:21.52 | tzafrir | Author: mdavenport not defined in AUTHORS file |
| 17:22.11 | tzafrir | (in repotools - this breaks my git-svn mirroring) |
| 17:22.30 | tzafrir | (and yeah, we soon won't need this, but still) |
| 17:27.31 | skrusty | does anyone know if it's possible from within FastAGI to get the DIALEDTIME variable if the caller hangs up? seems to be there's no way (from within one fastagi) of knowing any result of Dial because the channel is considered dead. Am i right? |
| 17:38.15 | seanbright | mdavenport = Malcolm Davenport? |
| 17:38.22 | leifmadsen | yes |
| 17:39.17 | seanbright | tzafrir: updated |
| 17:41.05 | tzafrir | thanks |
| 17:51.11 | seanbright | now we just need to figure out why he has two svn usernames |
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| 18:21.45 | file | gtjoseph, it lives. |
| 18:25.00 | file | now I just have to remember where stuff is... |
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| 19:21.04 | X-Rob | https://issues.asterisk.org/jira/browse/ASTERISK-18614 |
| 19:21.37 | seanbright | she said it's cold outside and she hands me my raincoat |
| 19:35.39 | JunK-Y | mjordan: let me know when you'll know please. |
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| 19:43.56 | gtjoseph | file: cool. it has a full dev environment so it should be ready to compile. |
| 19:44.06 | file | it was almost ready |
| 19:44.28 | gtjoseph | what was it missing? |
| 19:44.36 | file | a thing. #vague |
| 19:44.47 | gtjoseph | :) |
| 19:44.54 | file | I have register built |
| 19:45.13 | *** part/#asterisk-dev kharwell (~kharwell@nat/digium/x-fxgivfdktourubyr) |
| 19:45.26 | gtjoseph | did you have to make any changes or did it just build? |
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| 19:45.41 | file | it just built |
| 19:45.48 | gtjoseph | that bides well |
| 19:45.51 | gtjoseph | bodes |
| 19:51.28 | file | recall, recall, recall |
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| 20:00.42 | X-Rob | file: So. rtp multicast. I want to set the codec when I use Originate. |
| 20:01.28 | X-Rob | because when I use Originate, the rtp channel is sending out in slin. |
| 20:02.08 | file | have you tried specify Codecs? |
| 20:04.05 | X-Rob | file: I wasn't aware that I could, anywhere. Point me to the docs plz? |
| 20:04.27 | file | is it an AMI Originate? |
| 20:05.33 | file | otherwise I don't think the others allow specifying it |
| 20:06.35 | X-Rob | it's AMI yes, |
| 20:06.48 | seanbright | there should be a header |
| 20:06.48 | X-Rob | goes and looks |
| 20:07.10 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Originate |
| 20:07.16 | file | Codecs - Comma-separated list of codecs to use for this call. |
| 20:07.22 | seanbright | yes. that. |
| 20:07.40 | X-Rob | file: I love you |
| 20:07.42 | seanbright | file: you know things. |
| 20:07.49 | seanbright | and stuff. |
| 20:08.09 | file | seanbright, correct! |
| 20:08.30 | file | gtjoseph, I've almost got it |
| 20:08.38 | gtjoseph | :) |
| 20:09.23 | file | rewards self with British chocolate |
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| 20:12.00 | file | I'm only on rebuild 5 |
| 20:13.20 | gtjoseph | how long is a build taking? |
| 20:13.41 | file | awhile. |
| 20:14.01 | file | a make -j4 only does so much |
| 20:14.17 | gtjoseph | yep |
| 20:14.45 | gtjoseph | i use distcc on my farm |
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| 20:15.39 | file | wet a towel and wrap it around your head - it'll muffle the signal! |
| 20:15.48 | skrusty | :) |
| 20:16.01 | skrusty | "get your ass to mars!" |
| 20:16.37 | tzafrir | Do the the various test_* modules have any actual dependency on something from dev-mode? |
| 20:16.51 | gtjoseph | I'm certainly not going to jam that extractor gadget up my nose though. |
| 20:17.38 | tzafrir | Anything bad happens if I build them (and include them in a separate binary package)? |
| 20:17.55 | tzafrir | (performance degragation is also bad) |
| 20:18.28 | X-Rob | file: It works. I fuckin' love you man |
| 20:18.48 | file | X-Rob, That'll be 18 credits. |
| 20:20.02 | GameGamer43 | file: do you accept Digium Poker Chips? |
| 20:20.10 | file | GameGamer43, I do! |
| 20:20.30 | file | I actually wanted to grab a literal ton of them and make some sort of thing with a picture frame |
| 20:20.39 | X-Rob | I need to send file some timtams. |
| 20:21.11 | GameGamer43 | file: you should also accept teammates, they are delicious |
| 20:21.53 | file | oic |
| 20:23.40 | tonyclewis | So FreePBX 13 supports sending both multicast and unicast at the same time in page groups now. YAAAAAA |
| 20:24.03 | file | but - can it make my dinner for me? |
| 20:24.21 | coreyfarrell | anyone who has a minute https://reviewboard.asterisk.org/r/4094/ and https://reviewboard.asterisk.org/r/4038/ can use reviews |
| 20:24.22 | X-Rob | Yes. As long as your dinner consists of Unicast AND multicast RTP streams. |
| 20:24.37 | file | It does not. |
| 20:24.57 | X-Rob | Well in that case no. |
| 20:26.49 | leifmadsen | heh |
| 20:27.12 | leifmadsen | I should have grabbed a bunch so that I could use them as an actual poker chip stack |
| 20:30.56 | mjordan | coreyfarrell: still digging myself out of e-mail, if I don't get to those two today I will try to hit them tomorrow |
| 20:31.44 | coreyfarrell | mjordan: awesome, I've already found a large number of reference leaks using those changes, would love to give everyone else the ability to do the same |
| 20:31.48 | tzafrir | coreyfarrell, regarding 4094: minor nit |
| 20:32.06 | tzafrir | other files from contrib seem to be installed from the main makefile |
| 20:32.20 | tzafrir | Not sure if it is good or bad |
| 20:32.31 | coreyfarrell | tzafrir: the problem with that is access to the REF_DEBUG setting is not available from the main Makefile |
| 20:32.50 | coreyfarrell | long term I think maybe it would be better if contrib/Makefile handled anything in contrib/ ? |
| 20:33.43 | tzafrir | Anyway, it is installed to the astdatadir and not explicitly to /var/lib/asterisk, so it's fine by me |
| 20:34.19 | coreyfarrell | yeah I would never use an explicit install path :) |
| 20:34.24 | tzafrir | I think contrib/ is an odd name :-) |
| 20:34.38 | coreyfarrell | well except where it is needed to deal with something like a kernel module :) |
| 20:35.15 | file | gtjoseph, I... have it? |
| 20:36.09 | tzafrir | But yes, that Makefile in contrib/ is probably a better place in the long run |
| 20:36.31 | file | gtjoseph, what's your email? |
| 20:44.52 | coreyfarrell | to any ARI experts: testsuite/tests/rest_api/* open an ARI WS connection to Asterisk to create stasis apps and such. this not being closed causes 30-40 leaks to be reported by REF_DEBUG. should I be trying to close the connection from testsuite or Asterisk shutdown? |
| 20:56.46 | mjordan | looks |
| 20:58.42 | mjordan | coreyfarrell: I'd put the blame on the testsuite. If it isn't closing the WebSocket connection prior to shutting down Asterisk, it's not surprising that things end up in an unhappy ref state |
| 20:59.01 | mjordan | AriBaseTestObject in ari.py should probably be closing the underlying websocket connection when it detects that the reactor should be stopped |
| 21:03.16 | coreyfarrell | hmm looks like AriBaseTestObject would be responsible for tracking the existence of a WS connection, and AriTestObject would need to close it before calling self.stop_reactor() |
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| 21:10.45 | mjordan | coreyfarrell: you can override stop_reactor in AriBestTestObject |
| 21:10.59 | mjordan | have it call its base class function (TestCase.stop_reactor) after it closes the connection. |
| 21:11.23 | coreyfarrell | ah, that's probably a better solution I'll update my review request :) |
| 21:11.23 | mjordan | That way if someone else writes another class that inherits from AriBaseTestObject - which is possible - the WS connection is always closed correctly |
| 21:11.52 | coreyfarrell | should I worry about something in the future having multiple WS connections? |
| 21:13.46 | coreyfarrell | also I'm not the best at python, to call the base function I would want: super(AriBaseTestObject, self).stop_reactor() ? |
| 21:16.07 | mjordan | yup |
| 21:16.59 | mjordan | super is a nice way of saying "whoever my parent class(es) are, call the best implementation of this method" |
| 21:17.52 | mjordan | ah |
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| 21:18.00 | mjordan | and no, I would not worry about having multiple WS connections. |
| 21:18.09 | mjordan | If that is ever implemented, whoever implements it will have to worry about it |
| 21:19.28 | coreyfarrell | review updated |
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| 22:34.22 | jgowdy | hey guys, I'm trying to work on https://issues.asterisk.org/jira/browse/ASTERISK-22436 |
| 22:34.45 | jgowdy | I'm trying to add a flag to sip_pvt to track that we're in an Invite: Replaces |
| 22:35.02 | jgowdy | but when I do, the masq'ed channel doesn't have the same flag |
| 22:35.16 | jgowdy | I'm thinking that channel masq copies the sip_pvt fields somewhere I haven't wired up |
| 22:35.31 | jgowdy | or maybe channel masq only copies the string fields or something |
| 22:35.57 | jgowdy | Is there a place where sip_pvt values are preserved for something like this? |
| 22:41.36 | *** part/#asterisk-dev kharwell (~kharwell@nat/digium/x-mrejiapiguoeftiv) |
| 22:43.25 | jgowdy | hmm ast_channel_tech_pvt_set(original, ast_channel_tech_pvt(clonechan)); |
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| 23:28.46 | jgowdy | awesome, found the bug |
| 23:32.44 | coreyfarrell | jgowdy: that's great! do you know how to run the testsuite? I only ask because chan_sip tends to hard to change without breaking.. |
| 23:33.12 | jgowdy | I'll check out running the test suite. I know how finicky chan_sip is, it's not my first bug fix in there :) |
| 23:33.38 | coreyfarrell | cool.. thanks for working on the issue |
| 23:33.47 | jgowdy | But this change is pretty straight forward, and is only in the Invite: Replaces code path |
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| 23:47.41 | gtjoseph | jsmith: The 2.3.5 pjproject rpm worked great. |
| 23:48.00 | jsmith | gtjoseph: Glad to hear it -- I'll push those as updates |
| 23:48.13 | gtjoseph | cool and thanks! |