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00:34.31 | *** 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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13:44.14 | jkroon | asterisk.org domain is down? |
13:44.30 | jkroon | seems the DNS servers under digium.com is no longer resolving? |
13:44.46 | sibiria | works fine for me |
13:45.10 | file | there's some wonky DNS stuff for some people |
13:45.13 | Samot | Working for me. |
13:45.16 | file | I've raised a ticket with IT |
13:46.50 | file | it's not even 8AM in Huntsville though, so it'll be a bit |
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14:38.42 | jkroon | thanks file - from my side it looks like whois asterisk.org is indicating three nameservers (NSX{1,2,3}.DIGIUM.COM) - however, none of the digium.com nameserver names resolve. |
14:39.11 | jkroon | the nameservers in whois digium.com also disagrees with host -t ns digium.com. |
14:39.59 | file | it was a registrar problem, it's slowly resolving itself |
14:40.14 | jkroon | ok cool. then i'll just wait. |
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16:24.52 | jsmith | waves |
16:25.04 | jsmith | It's been a while since I've hung out in here, so I'll say hi :-) |
16:27.24 | seanbright | well if it isn't the VP of... community something |
16:27.50 | seanbright | or "mr. fancy pants" as i like to call you |
16:27.53 | file | warbles |
16:28.12 | seanbright | i was in my hotel room the night of the all conference happy hour and was wondering if it was still going on |
16:28.33 | seanbright | so i glanced out the window and saw the bright orange pants and knew things were still in full swing |
16:34.06 | Samot | Oh please tell me is was file in bright orange pants and a white top hat. |
16:34.15 | Samot | Because, that would be brave. |
16:34.17 | seanbright | no, it was jsmith |
16:34.21 | file | jsmith had the pants |
16:34.22 | file | I had the hat |
16:34.38 | file | apparently I was known by some as "the guy with the hat" |
16:34.51 | seanbright | i had a white carnation in my lapel. it was all very romantic. |
16:35.18 | Samot | Better than the guy with the pants. |
16:35.40 | Samot | "Which guy?" |
16:36.04 | Samot | "They guy wearing the pants that make you question is he a road worker or an escaped con" |
16:36.54 | Samot | Did said pants have reflective panels? |
16:37.14 | Samot | I mean were they "night time" safe? |
16:38.30 | sibiria | or did they make him look like the local county jail was a cowboy short? |
16:44.49 | jsmith | Samot: Sorry, no reflective panels... |
16:45.33 | Samot | I think you missed out on a serious statement. |
16:45.49 | Samot | I'm not sure what that statement would have been. But those pants would have made it serious. |
16:46.52 | Samot | Naw, I'm kidding. No one takes anyone wearing orange pants casually as serious. |
16:47.24 | seanbright | i have no idea what is going on |
16:47.33 | seanbright | jsmith was wearing bright orange pants. end of story. |
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16:49.35 | Samot | Hey, I used to wear Hawaiian shirts to the office because the dress code said shirts had to have a collar. |
17:00.04 | igcewieling | I used to put a rubber chicken on top of the screen of any computer I worked on and refused to talk about it. |
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18:25.39 | bkervaski | Hey guys, Asterisk 16.latest, PJSIP, have any of you seen a 400 "Bad Request" with the error being "Missing required SDP header 'm='"? New to me. Just popped up on 16.latest |
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18:30.10 | file | you'd need to show the SIP request in question |
18:30.24 | file | but it seems to think that the SDP has no m= line |
18:31.42 | bkervaski | It doesn't, looking at the request. Just v,o,s,c,t .. no "m" |
18:32.15 | bkervaski | direct media related? |
18:32.45 | file | that means the SDP has no media streams, we require one |
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18:34.11 | file | unless you mean that we are sending an SDP without a media stream and getting back a 400 Bad Request |
18:36.28 | bkervaski | It goes out without the m header .. but I think it's UDP related on my development vm .. digging into it |
18:36.51 | bkervaski | .. as local sip calls do have =audio 50128 RTP/AVP 9 0 101 |
18:36.59 | bkervaski | m==audio 50128 RTP/AVP 9 0 101 |
18:37.23 | file | nods |
18:37.55 | file | so it doesn't appear in the output of "pjsip set logger on" in the SDP? you could bump up debug and see if anything sticks out if that's the case |
18:38.54 | bkervaski | Correct. |
18:39.06 | bkervaski | (will do) thanks |
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19:47.32 | Banana51 | What are the best SIP + RTP libraries available? I have a python application I want to run on a remote VPS, and I want it to relay call audio (both ways) without using WebRTC since it requires a browser window to be open. |
19:48.17 | Banana51 | I will connect to my asterisk server, which is actually on the same VPS |
19:56.24 | lagzilla | I don't know about best but google revealed https://github.com/AGProjects/python-sipsimple |
19:57.37 | Banana51 | thanks for looking into it. That library is based off PJSIP (python C bindings). I have got it working but unfortunately the API does not allow me to store audio in buffers, which I need since I will relay the audio |
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20:47.40 | bkervaski | .. codecs .. wasn't allowing any codecs .. doh! |
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21:59.41 | Banana51 | ? it allows codecs, but send/reads the data stream directly to capture/playback device and not buffers that I can access |
22:01.39 | Banana51 | edit: never mind |
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