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12:22.03Demon_VoIPAsterisk 15.1, Strange behavior when negotiating codecs. Packet 15: http://bit.ly/2yGZ4uJ
12:23.22Demon_VoIPWhy asterisk do reinvite to caller? dialplan doesn't have PJSIP_SEND_SESSION_REFRESH(). asymmetric_rtp_codec=yes
12:24.06Demon_VoIPand second. Ok, answer was Unsupported media. Why is it a reason to "bye"?
12:26.49filealready been changed in 15 branch, will be in next release
12:29.32Demon_VoIPfile, I suspect that the bug introduced recently? for me it was critical. You can give a link to the patch?
12:30.06fileit's been there since 15 itself afaik
12:30.13filehttps://gerrit.asterisk.org/#/c/6759/
12:31.57fileI think there may be other fixes, but I'm on my phone
12:31.58Demon_VoIPhmm.. i am watching this issue. It was closed before 15.1 released.  Ok. Thank you.
12:32.39fileThe change I gave is not in 15.1
12:32.54fileAnd do you have codecs allowed on an endpoint that it does not support?
12:33.27Demon_VoIPi see changelog: 2017-09-25 12:30 +0000 [3d658e3be4]  Kevin Harwell <kharwell@digium.com>  ASTERISK-27259 #close
12:34.16Demon_VoIPyes. endpoint config allows a lot of condecs. And devices are not (of course)
12:34.19fileYes, the change I gave came after.
12:34.38Demon_VoIPfile, thank you. i'll try to patch and test
12:35.24fileThe code is trying to better match the call codecs. Since you configured it to say it supports the codec, it sent a reinvite out.
12:38.23Demon_VoIPHope it will not be necessary for each endpoint to fill to "allow" only available on device codecs?
12:39.21fileThe patch makes it so the already negotiated ones are included and the reinvite will succeed so it's not required to only allow what they support
12:39.34fileBut if it was configured to match the reinvite would never occur
13:06.51Demon_VoIP15.1 one or more time by day segfaults on codec_opus.c: Opus: failed to create encoder (-1)
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13:08.07Demon_VoIPNov  8 11:48:47 talk37 kernel: [8283436.975308] asterisk[20299]: segfault at 60 ip 00007faf239b27a2 sp 00007faef0a4f410 error 4 in codec_opus.so[7faf239a4000+54000]
13:08.07Demon_VoIPNov  8 14:49:15 talk37 kernel: [8294265.061166] asterisk[23345]: segfault at 60 ip 00007f2f3ef877a2 sp 00007f2f0b14a420 error 4 in codec_opus.so[7f2f3ef79000+54000]
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13:09.55fileThere is an issue filed by Torrey which is that
13:10.23Demon_VoIPtwo days ago i can't find it :(
13:10.42fileJust search for codec_opus, it's there
13:12.04filehttps://issues.asterisk.org/jira/browse/ASTERISK-27381
13:13.52Demon_VoIP24 issues.. Thank you! I found only this: https://issues.asterisk.org/jira/browse/ASTERISK-26810
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13:15.18Demon_VoIPbut i don't use WEBRTC. i hope it doesn't matter
13:15.47fileOpus is opus.
13:19.52Demon_VoIPafter the entry into service the gateway in the telegram I made opus the default codec and widely used. Because it becomes a problem :(
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13:24.42voipmonkmmm, a demon...
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14:02.11iheartlinuxanyone know how to make the sip (pjsip) address different from the username on an obi200?
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14:08.26SamotExpalin.
14:08.31SamotExplain
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14:16.46Samotiheartlinux: You need to clarify your question.
14:33.21iheartlinuxSamot: PJSIP (AuthId/UserName), the obi200 only has a username field
14:33.53SamotOK.
14:33.54[TK]D-FenderWhy did you say "address"?
14:34.00SamotThen that will be used for that authid
14:34.07SamotThat's how it works.
14:34.18[TK]D-FenderAnd most devices only use a single name anyway
14:34.46iheartlinuxuse to polycom, it has two fields for authid/username, namely address & authenication user id
14:34.55SamotOK
14:34.58SamotNot all phones have that
14:35.13SamotThe username will be used as the authid if the auth field doesn't exist.
14:37.28iheartlinuxtnx
14:37.42SamotAlso..
14:37.52SamotThere is an auth= setting in PJSIP for this.
14:38.11SamotOh wait..
14:38.19SamotThat is for the [auth] section.
14:40.03iheartlinuxI try to keep them different where possible, but I use an obscure port with pjsip, so it's not a big deal
14:40.31SamotThere's no need to keep them separate.
14:40.42SamotIn fact that's not even a practice really anymore.
14:40.53iheartlinuxorly
14:41.51SamotThe authid was meant as a secondary measure.
14:41.57SamotI give you a SIP user/pass
14:42.10SamotAnd also say "put the user in the auth"
14:42.23SamotIf someone gets your SIP user/pass but doesn't know the auth is needed..
14:42.31lvlinuxActually, it is a practice with pjsip, so that on a public facing server you can can have a phone register to a stupid extension/address like "101" and have the auth user be something complex, so that brute force attempts won't work on "101".
14:42.37SamotThen when they try to register or auth and it's missing...boom done.
14:42.40SamotCan't do anything.
14:42.54SamotWhat?
14:43.02SamotPJSIP has nothing to do with that
14:43.04SamotSIP is SIP
14:43.10SamotPJSIP is just a driver.
14:43.18lvlinuxYes, but PJSIP makes doing it that way more practical.
14:43.24SamotNo.
14:43.30SamotThat has nothing to do with it.
14:43.40SamotYou're case is specific.
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14:43.52SamotDoesn't apply to anything that is not over the Internet.
14:43.56SamotAnd also..
14:44.03SamotITSPs don't care if you use PJSIP
14:44.15SamotNo one on the other side cares if you are using PJSIP
14:44.20SamotThat doesn't matter to them, at all.
14:44.24lvlinuxwell of course not
14:44.33lvlinuxi just gave that as an example...
14:44.44SamotSo they aren't giving you creds that work under this scenario.
14:44.55SamotThat aren't taking what you pointed out into consideration.
14:45.20SamotHaving a different auth vs user name isn't going to protect you from a brute force.
14:45.34SamotThat isn't even on the table for stopping that kind of attack.
14:46.14lvlinuxno, it doesn't protect you, but it keeps the noise down in the logs, and lets you pick a more lengthy username without sacrificing a simple endpoint name.
14:47.04lvlinuxi take that back, it won't keep any noise down in the logs I guess. But the rest is true.
14:47.29SamotThat's what a firewall is for.
14:47.49SamotIf your main mode of security if username and auth are different..
14:47.55SamotThen you are already failing.
14:48.13lvlinuxOf course, but it's one part of the system. To depend solely on a firewall is naive.
14:48.24SamotYes..
14:48.25SamotBut..
14:48.29lvlinuxWithout makeing sure the whole config is relatively secure as a whole.
14:48.42SamotThis is kinda what I do.
14:48.44SamotI know.
14:48.52SamotI build these things all the time.
14:48.56lvlinuxYes of course.
14:49.21SamotI've had this discussion with providers.
14:49.38SamotWhen they've brought me in to rebuild/fix their networks.
14:49.50SamotGenerally after they've been hacked numerous times.
14:50.00SamotBecause they employed 2005 Asterisk security rules.
14:51.09lvlinuxUsing a simple user and complex auth_user isn't a magical security fix, it's a simple tactic that just makes administration of a [usually a small] system easier without sacrificing any security.
14:51.32SamotOK..
14:51.41SamotThen you realize that you have to store that in two places right?
14:52.02SamotBecause the auth user only exists and is presented at the time of auth.
14:52.20SamotThe rest of the time the user aka FROM user is used for everything else.
14:52.30SamotSo since that auth user will be different..
14:52.32lvlinuxYes.
14:52.39SamotYou have to base your lookups in two different places.
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14:53.37iheartlinuxdidn't mean to cause a war :)
14:54.03SamotThat and most REGISTRAR/Location systems consider mismatching user/auth ids to be a warning/error
14:55.36lvlinuxBut then we're talking about something besides Asterisk, aren't we? If you do that, then of course what I said goes out the window. :)
14:56.13SamotYes but Chan_SIP did the same thing.
14:56.26SamotSo this wasn't a magical PJSIP thing you claimed it to be.
14:56.28Samotauthuser
14:56.29lvlinuxiheartlinux: big diff between a "war" and a "healthy discussion" :)
14:56.31Samotdefaultuser
14:56.34Samotfromuser
14:56.40SamotAll settings in Chan_SIP
14:57.00SamotThe register string had a spot for both the username and the auth user
14:57.16SamotSo this idea that PJSIP brought something new to the table with that is wrong.
14:58.16iheartlinuxJust exaggerated a little :)
14:58.30lvlinux:-D
14:58.43SamotAnd yes.
14:58.54SamotWhen I talk about Telephony, SIP and how these things play together..
14:59.02SamotI talk about more than just Asterisk.
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14:59.06SamotBecause Asterisk is just a subset
14:59.35SamotIt leads people to do things like ask ITSPs and carriers "Do you support PJSIP"
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14:59.43lvlinuxlol
14:59.47SamotOr think that somehow PJSIP has magically changed how SIP works.
15:00.05lvlinuxNo it hasn't, but it has simplified config in some ways.
15:00.25SamotSure.
15:00.46SamotBut PJSIP is a late comer.
15:01.00SamotThe party started at 8 and they showed up a 1am
15:01.23lvlinuxat least they showed up :)
15:03.05iheartlinuxI'm glad it showed up. I use both chans, sip for carrier, pjsip for devices
15:03.10SamotSo back to what I said...
15:03.26SamotIf the device does not have a Auth Username field or option.
15:03.41SamotWhen auth happens the username name is pushed to the auth field.
15:04.07SamotBecause of that, 99.99% of the time the username/authuser are the same.
15:04.24SamotDue to the lack of support by _devices_ and _softphones_
15:04.59iheartlinuxtrue
15:05.43lvlinuxyes
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18:17.50*** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.1 (2017/11/08), 11.25.3 (2017/09/19), Standard: 15.1.1 (2017/11/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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18:36.33jamesaxlhello
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18:56.03Samotfile: How long will 11.25.3 stay in the LTS section of the topic?
18:57.49[TK]D-FenderShould say NMS
18:58.08[TK]D-FenderOr debateably disappear
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19:27.18*** topic/#asterisk by cresl1n -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.1 (2017/11/08), Standard: 15.1.1 (2017/11/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
19:27.25cresl1nSamot: there you go
19:27.40SamotWOOO
19:27.48SamotI IMPACTED A CHANGE
19:27.55SamotI AM SO FOSS!
19:29.21lvlinux:-D
19:30.18SamotThe next time someone says "Digium doesn't take feature requests"
19:30.28SamotI can now say "Well maybe not from *you*.."
19:31.31SamotI'm going to AsteriCon next year with a shirt that says "cresl1n accepted my feature request. What about yours?"
19:31.40MrTAPThat wasn't a feature request, was a bug report.
19:31.56SamotYou're not raining on my parade.
19:32.01SamotNot happening.
19:32.08SamotAin't no one got time for that
19:32.44cresl1nSamot: were you at Astricon this year?
19:32.52SamotNo.
19:33.19SamotBut now..
19:33.38SamotI just might have to be next year so people can hail me for my achievement.
19:33.54SamotI will dress in god like robes and naked women will throw little pickles at me.
19:46.21[TK]D-FenderSamot, https://www.youtube.com/watch?v=9VDvgL58h_Y
19:46.56cresl1nSamot: It'd be great to see you there next year
19:47.34Samothttps://www.youtube.com/watch?v=SgNgAiGQ_IM
19:50.00avbi want to attend astricon next year too
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21:11.19cervajs2any ideas why res_rtp_asterisk.so: undefined symbol: DTLS_method with asterisk 15.0.0 and libsrtp 1.5.4 ?
21:13.23cervajs2yup. yum update openssl solved the problem
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21:14.05cervajs2road to Asterisk SFU clean
21:15.31cresl1n?
21:22.10cervajs2:) testing asterisk 15 as SFU. works nicely
21:25.52cervajs2for the record: i must update from openssl-1.0.1e-60.el7_3.1.x86_64 to openssl-1.0.2k-8.el7.x86_64 . updated my spec file for minimal openssl version
21:26.46cresl1nNice!
21:26.56cresl1nAre you using the client up on github?
21:27.12cresl1ncybermegaphone2k?
21:27.17cervajs2yes
21:30.06cresl1nThat's great
21:30.06cervajs2cresl1n: are u using libsrtp 2.x ?
21:30.14cresl1nI just installed it
21:30.22cresl1nhaven't tested it yet with anything significant
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21:40.43cervajs2chm its not working on android/chrome
21:41.20cervajs2ops. audio from mobile is working. no video
21:46.52cresl1nwe haven't don't done qualification with mobile clients
21:47.00cresl1nit's been pretty much all browser clients thus far
21:47.22cresl1nsorry, desktop browser clients
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23:03.59startledmarmot@cresl1n: we're actually going to be deploying a 15-based SFU at a client soon, so :thumbsup:
23:04.17cresl1nWhat!
23:04.20cresl1nThat's awesome!
23:04.21startledmarmotI know, right
23:04.23startledmarmot;)
23:04.30cresl1nCan I see it?
23:04.33cresl1n:-D
23:04.50startledmarmotI think that can be arranged when it's all up and runnin'!
23:04.58cresl1nthat'd be awesome, if you don't mind
23:05.02startledmarmotcan I trade you for some trade secrets on getting the screensharing working?
23:05.13startledmarmotbut for sure man
23:05.35cresl1nAre you having trouble with screen sharing?
23:05.44cresl1nAlso, startledmarmot what's your name IRL?
23:06.00startledmarmotEvan McGee
23:06.05cresl1nOH
23:06.06cresl1nI KNOW YOU
23:06.08startledmarmothahaha
23:06.28startledmarmotusually hiding behind the bowties :P
23:06.34cresl1nI like the bowties
23:06.38cresl1nthey're always in good taste
23:07.04startledmarmotaww, thanks dude. they do bring the requisite whimsy.
23:07.20startledmarmoton the screen sharing front, we haven't dug in yet
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