IRC log for #asterisk on 20161005

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01:09.39monstercoI know there was a page that I could use to test ARI functions on my server by pointing it to my server - I have forgotten what it was. Can anyone link me please?
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02:54.25lemmy2010Hello, anybody still hanging out and talking Asterisk?
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03:00.54SamotOnly if it's about Asterisk 1.6.
03:01.00SamotAnd how awesome of a release it was.
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03:21.00lemmy2010I thought all new releases were great?
03:53.51drmessano1.6 was kinda the apex
03:54.31drmessanoAfter that it was basically Google Voice and web based calling nonsense
03:54.35drmessanoIt all went to hell
03:58.31drmessano"Can I WebRTC with it from my Apple Wireless Headphones?" <-- Probably an average sales call to Digium now
03:58.33drmessanoSo sad
04:05.13Samotrolf.
04:06.02*** join/#asterisk Alex_Bkash (671a7011@gateway/web/freenode/ip.103.26.112.17)
04:06.02SamotiOS 10 probably removed an important feature that WebRTC needs.
04:06.27drmessano"Why doesnt Asterisk auto-complete my hashtags?"
04:06.39SamotWelcome to iSIP
04:07.22SamotYou can stream MoH straight from iTunes.
04:07.40drmessano15 will probably have Emoji
04:07.47drmessanoBecause it's requested
04:07.53SamotUse the video camera to auth your calls by retina.
04:08.20Samot"I can't make an outbound call"
04:08.24Samot"Clean your lens"
04:08.32drmessano"Wrong Eye, dipshit"
04:09.25drmessanoI'll still love Asterisk even after it has Snapchat integration
04:09.33drmessanoBut just sad
04:10.03Samot"Press *43ll to Vine now"
04:10.28drmessanoMy home PBX does have its own Twitter account
04:11.04SamotI wonder if I can make my VVX600 swipe left to deny a call and swipe right to accept.
04:11.35drmessanoPress # to send a Wiener Pic
04:11.40drmessano#2016
04:13.07SamotAnd you'll still have GV on Android.
04:14.06SamotBut at least you'll have a headphone jack.
04:14.57drmessanoFor now
04:15.50SamotOh I'm sure you'll be looked down on by hipsters and millen-douches for have a headphone jack.
04:16.29SamotAnd in five years they'll be scoffing you for not be cool and retro for *not* using a headphone jack.
04:16.42SamotYou'll just never win
04:17.09drmessanoI tried to drink a craft beer
04:17.27drmessanoI assumed they meant "Kraft"
04:17.27SamotDid your tastebuds hate you for it?
04:17.42drmessanoSo I put a half pound of Cheez Wiz in it
04:17.55drmessanoBefore it curdled it wasnt half bad
04:19.01*** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net)
04:20.33SamotOh..
04:20.43SamotYou should check out Nita Struass.
04:20.58SamotOr is it Strauss? I think it's Strauss.
04:21.29SamotIf you're into women that can make a guitar their bitch.
04:23.08lemmy2010What if I just want to understand how a non-hipster feature might work?
04:23.25SamotThose don't exist on iPhones.
04:24.20drmessanolemmy2010: That's all fine and good if you only want to make calls
04:24.25drmessanoWho does that anymore? pffft
04:24.35lemmy2010True, not me
04:25.12lemmy2010I'm trying to understand SMDI
04:28.01SamotHrm, I wonder how many "cool points" I'll get if I turn up a Minecraft server for the kids.
04:29.11drmessanoUse Spigot
04:30.20SamotI was thinking about doing it as a Christmas gift for them.
04:30.29drmessanoAwesome
04:31.02SamotBecause when you have twins you try to find the things that can be "shared"
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05:04.07*** join/#asterisk iheartlinux (~jwpierce3@mail.johnthecomputerguy.com)
05:04.44iheartlinuxis it possible to run multiple cmds on single line in extensions.conf?
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05:06.28drmessanoYou can nest some commands
05:06.33drmessanoDepends what youre trying to do
05:11.00iheartlinuxI defined a context for each user. I'm calling each context -> phone/device with goto, so for readability, I wanted to append hangup to each Dial or Queue lines
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05:36.19drmessanookay
05:41.58iheartlinuxit's not a big deal. I am using "same n," just figured I'd ask
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05:53.25drmessanoStuff like string manipulation can be nested, but as far as commands line Dial and Hangup, no
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06:34.27*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.2 (2016/09/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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13:24.52seik0Hi! is it true, that Asterisk in chan_sip treats incoming packets  "183 Session Progress" as "180 Ringing" (at least, in 11 version) ?
13:25.14*** join/#asterisk stefan27 (~stefan27@static-212-247-4-149.cust.tele2.se)
13:27.49seik0ok, looks it is true: https://issues.asterisk.org/jira/browse/ASTERISK-18987
13:29.37SamotAre you sure?
13:29.40SamotThat's a bug from 1.8
13:29.59SamotAre you sure releases after that didn't have a fix for this bug?
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13:30.39seik0that's a bug coming from 1.4
13:31.20seik0and somewhere i see a "patching instruction" for 11
13:31.44seik0it is, of course, possible, that it solved later
13:31.45stefan27Hmm, I have a special asterisk 13.5.0, using chan_sip and setup on a public IP with empty [default] context. When some bad guy spams invites like "INVITE yyy@mydomain.com , From: xxx@mydomain.com" with randomyzed xxx, yyy, asterisk initially always responds with 404 Not Found, but after ~1000 invites it seems to start responding 403 Forbidden
13:31.54stefan27does anyone have a clue why?
13:32.03[TK]D-FenderNobody has talked about this ticket at all
13:32.04[TK]D-Fender<PROTECTED>
13:32.35*** join/#asterisk anonymouz666_ (bd193d81@gateway/web/freenode/ip.189.25.61.129)
13:32.39[TK]D-Fenderstefan27, Show us
13:32.53anonymouz666_hi all
13:33.08anonymouz666_what's the equivalent of insecure=port,invite in chan_pjsip?
13:33.31seik0[TK]D-Fender:  do not complete understand what  this link means
13:34.01Samotseik0: It's an internal link, we can't view it.
13:34.02[TK]D-Fenderit means there is ANOTHER ticket# with related followup
13:34.22Samot^ That.
13:34.35seik0another closed for publicity ticket?
13:34.40SamotBut it's internal to Digium. Which means the developers were already working on the bug
13:34.48SamotI  would assume.
13:35.15SamotAgain, you're looking at a bug from like 4 major releases ago.
13:35.24seik0just said, that nobody has talked about it at all?
13:35.32seik0or about this ticket?
13:35.40SamotBecause it was taken internal.
13:35.50seik0but bug is not closed, not marked as resolved in later releases
13:36.06SamotBecause someone didn't close that ticket.
13:36.11SamotWhy? I don't know.
13:36.18seik0as linked it should have been closed just when resolved?
13:36.26SamotBut SWP-4043 is the ticket that issue is *really* under.
13:37.18Samotseik0: I don't know.
13:37.37SamotI don't know internal Digium policies on handling tickets. So why speculate.
13:37.40stefan27I can't reproduce it, all I have is a wireshark pcap file which also contains sensitive customer data so it's not uploadable. The extensions.conf is just empty [default] and none of the invites from the bad guy has any SRC IP or from user that matches a sip entity.
13:38.07stefan27I'm just curious if anyone with knowledge of asterisk and chan_sip could *guess* why asterisk suddenly starts responding 403 instead of 404
13:39.39eric_hillstefan27: One of the inbound SIP invites gets assigned a call ID that was already used by a legitimate call and asterisk finds it in an internal auth table. </guessing>
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13:40.47seik0stefan27,  depends on configuration of sip users, theirs names, theirs password, theirs insecure settings, nat settings. I guess.
13:40.51eric_hillstefan27: Asterisk maintains several internal tables for active calls, and I'm betting the call from your pcap matches a prior call that you might be lucky to find in cdr-csv </guessing>
13:42.21[TK]D-Fenderstefan27, maybe things aren't what you think they are and you'r asking us to guess on assumed evidence.  There is no way in this universe that I'm going to play that game.
13:42.47seik0stefan27: on incoming calls asterisk tries to link authentication data with existin sip accounts in asterisk and different partial conicidens result in different errors
13:43.24seik0I guess, again
13:43.59eric_hillActually, cosmic radiation could be a good guess too.
13:47.32seik0In 13.11.2 183 Alcatel workaround is present, so noone has talked is just means talk about whole problem
13:48.09stefan27thanks, I'm happy for the guesses; I too have insufficient evidence of what really happend; so the guessing game is the best I can do
13:49.26seik0off, wrting patch
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21:59.48*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.2 (2016/09/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
22:06.27iheartlinuxexit
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