IRC log for #asterisk on 20191111

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06:06.16*** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.1 (2019/10/16) 16.6.1 (2019/10/16) Standard: 17.0.0 (2019/10/28); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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13:36.45s3c70rhi
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14:23.39s3c70rLately i ran into some type of bug/race condition which caused Asterisk to SIGSEGV may someone help me out to locate/fix the bug?
14:24.24filewhat version of Asterisk?
14:24.25SamotDescribe the actual issue.
14:24.29SamotAnd that.
14:24.58s3c70ri can tell you in one second
14:27.00s3c70rVersion is 13.21
14:28.37s3c70rThe server is heavily loaded and randomly crashes by NULL pointer dereference of tsx->transport
14:30.10fileyour version of Asterisk and of PJSIP is old, while someone could try to chase such a thing down it may already be fixed
14:31.15s3c70rthe problem is i have to work with that and it wont be an option for me to port everything to a newer version
14:34.14s3c70rAlso it is asterisk certified version and there does not exist any newer certified version or may i be wrong here?
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14:35.17filethere is no newer version of certified currently
14:45.58s3c70ralso i noticed while analysis that some earlier stack frames which are close to the outermost frames got overwritten by some type of continuous pattern looking like some pointer arithmetic overwriting ioqueue pool structures. The problem is i was not capable of pointing out the real cause since asterisk is a highly multi threaded object.
14:47.15s3c70rso the actual cause of the segfault was in /src/pjsip/sip_transaction.c:1884 where PJSIP_TRANSPORT_IS_RELIABLE(tsx->transport) dereferences a NULL pointer.
14:49.40s3c70rall crashed occurred while ioqueue holds an read_event and the actual buffer of resolvers TX and RX does not contain any data which was going in ~8h before the crash telling me the data had to stay in the queue quit long before the crash happened
14:50.30Samots3c70r: Do you have the actual support contract that goes with Certified or did you just use that version?
14:53.41s3c70rwe use the open source version without any contract
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16:02.01s3c70rhello?
16:04.55SamotThe suggestion is that you update
16:05.01SamotTo the non certified version.
16:09.40s3c70rI thought the certified version should be more stable than the newer ones and getting support is essential especially on this sort of problems.
16:13.26[TK]D-FenderOnly more supported by direct customers with contracts
16:29.42SamotInstead of getting updated every 4-6 weeks it's updated 4 times a year
16:34.22s3c70rthx
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18:57.11igcewielingI  am unable to get the value of DIALSTATUS from inside a hangup handler.  CLI output of a test call using a simplified dialplan.  https://pastebin.com/zb4bitSZ
18:57.37igcewielingI'm hoping I'm missing something and not that it isn't supported.
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18:58.36fileDIALSTATUS is set on the calling channel
18:58.53fileyou're executing on the called channel
18:58.56igcewielingfile: HANGUPCAUSE is accessable.
18:59.14filesure, that's for the channel itself
18:59.15SamotYes but the called channel didn't Dial()
18:59.30igcewielingI also can't access is from MASTER_CHANNEL(DIALSTATUS) which should be the calling channel.
19:00.10fileit's probably executing before DIALSTATUS has been set
19:00.33igcewielingah, I'll add a wait and see what happens.
19:01.28fileyour logging doesn't include the thread ID, so I can't see where it is executing - whether it is acting synchronous or asynchronous
19:03.56igcewielingIt will be easier to map a hangupcause to a dialstatus than it would be to try to access the value of DIALSTATUS in the parent channel.
19:04.28igcewielingAdding a wait generates [2019-11-11 14:04:06] WARNING[18683][C-000000b0]: channel.c:4864 ast_prod: Prodding channel 'PJSIP/vzb-regular-00000180' failed
19:05.24igcewielingthere is an RFC for the mappings.
19:06.12igcewielingor better yet, use the SIP cause.
19:08.11igcewielingI'm trying to generate a CEL event with hangup info from the hangup handler independent of an agi or other dialplan.  Thanks for the info file.
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23:04.04Samot@file: Got a question for you.
23:05.14SamotIf I Page() a group of endpoints (all PJSIP except one Local) and that Local is setup to Page() a multicast stream...what would happen?
23:05.41fileit'd call all the PJSIPs and also do multicast
23:05.50drmessanoOoooh
23:05.59SamotSo the Local would bridge up and still stream through?
23:06.05fileyes
23:06.11drmessanoFsck yes
23:06.15Samotdrmessano: There you go.
23:06.21SamotMy theory was correct.
23:07.20Samot6:00:31 PM <drmessano> Page ---> 101, 102, 103, 999
23:07.20Samot6:00:42 PM <drmessano> 999 Custom Extension---> Local/page-extension
23:07.20Samot6:01:27 PM <drmessano> page-extension -----> Page(MulticastRTP/linksys/239.10.1.1:5004/239.10.1.1:6061)
23:07.30SamotSo that flow works?
23:07.32SamotSweet.
23:07.38drmessanofile: I want to call a MulticastRTP page to say go to no-phone sets, but include it in one of the FreePBX GUI page groups.. So the idea was to Russian Nesting Doll it..
23:07.52filesure
23:08.09drmessanoThat kicks ass.. Ok, then I will get cracking on it
23:08.14drmessanoThanks
23:08.22Samot❤️ ✳️
23:09.26drmessanoSamot: Oh and I want to route the Multicast across a couple MT boxes, because the PBX is up in a cloud
23:09.37drmessanoSO multiple points of fom tuckery to play with
23:09.44SamotYeah , let me know how that goes
23:09.51SamotBecause that should be interesting.
23:10.00drmessanoI actually THINK I got the Multicast routing working
23:10.13drmessanoOnce I get home I am gonna start the rest
23:10.57Samothttps://pics.onsizzle.com/multi-rass-bitch-memeful-com-13561174.png
23:11.36SamotI don't care what anyone says. I f'ing love that movie.
23:12.53drmessanoI pushed the MulticastRTP config to my SPA devices at home.  I can see recievers joined on 239.10.1.1 from my home <> CHR
23:13.02drmessanoSo on paper, so far so good
23:13.25drmessanoHeaded home now to try a page
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