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01:09.39 | monsterco | I 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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02:54.25 | lemmy2010 | Hello, anybody still hanging out and talking Asterisk? |
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03:00.54 | Samot | Only if it's about Asterisk 1.6. |
03:01.00 | Samot | And how awesome of a release it was. |
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03:21.00 | lemmy2010 | I thought all new releases were great? |
03:53.51 | drmessano | 1.6 was kinda the apex |
03:54.31 | drmessano | After that it was basically Google Voice and web based calling nonsense |
03:54.35 | drmessano | It all went to hell |
03:58.31 | drmessano | "Can I WebRTC with it from my Apple Wireless Headphones?" <-- Probably an average sales call to Digium now |
03:58.33 | drmessano | So sad |
04:05.13 | Samot | rolf. |
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04:06.02 | Samot | iOS 10 probably removed an important feature that WebRTC needs. |
04:06.27 | drmessano | "Why doesnt Asterisk auto-complete my hashtags?" |
04:06.39 | Samot | Welcome to iSIP |
04:07.22 | Samot | You can stream MoH straight from iTunes. |
04:07.40 | drmessano | 15 will probably have Emoji |
04:07.47 | drmessano | Because it's requested |
04:07.53 | Samot | Use the video camera to auth your calls by retina. |
04:08.20 | Samot | "I can't make an outbound call" |
04:08.24 | Samot | "Clean your lens" |
04:08.32 | drmessano | "Wrong Eye, dipshit" |
04:09.25 | drmessano | I'll still love Asterisk even after it has Snapchat integration |
04:09.33 | drmessano | But just sad |
04:10.03 | Samot | "Press *43ll to Vine now" |
04:10.28 | drmessano | My home PBX does have its own Twitter account |
04:11.04 | Samot | I wonder if I can make my VVX600 swipe left to deny a call and swipe right to accept. |
04:11.35 | drmessano | Press # to send a Wiener Pic |
04:11.40 | drmessano | #2016 |
04:13.07 | Samot | And you'll still have GV on Android. |
04:14.06 | Samot | But at least you'll have a headphone jack. |
04:14.57 | drmessano | For now |
04:15.50 | Samot | Oh I'm sure you'll be looked down on by hipsters and millen-douches for have a headphone jack. |
04:16.29 | Samot | And in five years they'll be scoffing you for not be cool and retro for *not* using a headphone jack. |
04:16.42 | Samot | You'll just never win |
04:17.09 | drmessano | I tried to drink a craft beer |
04:17.27 | drmessano | I assumed they meant "Kraft" |
04:17.27 | Samot | Did your tastebuds hate you for it? |
04:17.42 | drmessano | So I put a half pound of Cheez Wiz in it |
04:17.55 | drmessano | Before it curdled it wasnt half bad |
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04:20.33 | Samot | Oh.. |
04:20.43 | Samot | You should check out Nita Struass. |
04:20.58 | Samot | Or is it Strauss? I think it's Strauss. |
04:21.29 | Samot | If you're into women that can make a guitar their bitch. |
04:23.08 | lemmy2010 | What if I just want to understand how a non-hipster feature might work? |
04:23.25 | Samot | Those don't exist on iPhones. |
04:24.20 | drmessano | lemmy2010: That's all fine and good if you only want to make calls |
04:24.25 | drmessano | Who does that anymore? pffft |
04:24.35 | lemmy2010 | True, not me |
04:25.12 | lemmy2010 | I'm trying to understand SMDI |
04:28.01 | Samot | Hrm, I wonder how many "cool points" I'll get if I turn up a Minecraft server for the kids. |
04:29.11 | drmessano | Use Spigot |
04:30.20 | Samot | I was thinking about doing it as a Christmas gift for them. |
04:30.29 | drmessano | Awesome |
04:31.02 | Samot | Because when you have twins you try to find the things that can be "shared" |
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05:04.44 | iheartlinux | is it possible to run multiple cmds on single line in extensions.conf? |
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05:06.28 | drmessano | You can nest some commands |
05:06.33 | drmessano | Depends what youre trying to do |
05:11.00 | iheartlinux | I 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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05:36.19 | drmessano | okay |
05:41.58 | iheartlinux | it's not a big deal. I am using "same n," just figured I'd ask |
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05:53.25 | drmessano | Stuff 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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13:24.52 | seik0 | Hi! 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) |
13:27.49 | seik0 | ok, looks it is true: https://issues.asterisk.org/jira/browse/ASTERISK-18987 |
13:29.37 | Samot | Are you sure? |
13:29.40 | Samot | That's a bug from 1.8 |
13:29.59 | Samot | Are you sure releases after that didn't have a fix for this bug? |
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13:30.39 | seik0 | that's a bug coming from 1.4 |
13:31.20 | seik0 | and somewhere i see a "patching instruction" for 11 |
13:31.44 | seik0 | it is, of course, possible, that it solved later |
13:31.45 | stefan27 | Hmm, 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 |
13:31.54 | stefan27 | does anyone have a clue why? |
13:32.03 | [TK]D-Fender | Nobody has talked about this ticket at all |
13:32.04 | [TK]D-Fender | <PROTECTED> |
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13:32.39 | [TK]D-Fender | stefan27, Show us |
13:32.53 | anonymouz666_ | hi all |
13:33.08 | anonymouz666_ | what's the equivalent of insecure=port,invite in chan_pjsip? |
13:33.31 | seik0 | [TK]D-Fender: do not complete understand what this link means |
13:34.01 | Samot | seik0: It's an internal link, we can't view it. |
13:34.02 | [TK]D-Fender | it means there is ANOTHER ticket# with related followup |
13:34.22 | Samot | ^ That. |
13:34.35 | seik0 | another closed for publicity ticket? |
13:34.40 | Samot | But it's internal to Digium. Which means the developers were already working on the bug |
13:34.48 | Samot | I would assume. |
13:35.15 | Samot | Again, you're looking at a bug from like 4 major releases ago. |
13:35.24 | seik0 | just said, that nobody has talked about it at all? |
13:35.32 | seik0 | or about this ticket? |
13:35.40 | Samot | Because it was taken internal. |
13:35.50 | seik0 | but bug is not closed, not marked as resolved in later releases |
13:36.06 | Samot | Because someone didn't close that ticket. |
13:36.11 | Samot | Why? I don't know. |
13:36.18 | seik0 | as linked it should have been closed just when resolved? |
13:36.26 | Samot | But SWP-4043 is the ticket that issue is *really* under. |
13:37.18 | Samot | seik0: I don't know. |
13:37.37 | Samot | I don't know internal Digium policies on handling tickets. So why speculate. |
13:37.40 | stefan27 | I 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. |
13:38.07 | stefan27 | I'm just curious if anyone with knowledge of asterisk and chan_sip could *guess* why asterisk suddenly starts responding 403 instead of 404 |
13:39.39 | eric_hill | stefan27: 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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13:40.47 | seik0 | stefan27, depends on configuration of sip users, theirs names, theirs password, theirs insecure settings, nat settings. I guess. |
13:40.51 | eric_hill | stefan27: 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-Fender | stefan27, 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. |
13:42.47 | seik0 | stefan27: on incoming calls asterisk tries to link authentication data with existin sip accounts in asterisk and different partial conicidens result in different errors |
13:43.24 | seik0 | I guess, again |
13:43.59 | eric_hill | Actually, cosmic radiation could be a good guess too. |
13:47.32 | seik0 | In 13.11.2 183 Alcatel workaround is present, so noone has talked is just means talk about whole problem |
13:48.09 | stefan27 | thanks, 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 |
13:49.26 | seik0 | off, 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 |
22:06.27 | iheartlinux | exit |
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