IRC log for #asterisk on 20191114

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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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13:44.14jkroonasterisk.org domain is down?
13:44.30jkroonseems the DNS servers under digium.com is no longer resolving?
13:44.46sibiriaworks fine for me
13:45.10filethere's some wonky DNS stuff for some people
13:45.13SamotWorking for me.
13:45.16fileI've raised a ticket with IT
13:46.50fileit's not even 8AM in Huntsville though, so it'll be a bit
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14:38.42jkroonthanks 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.
14:39.11jkroonthe nameservers in whois digium.com also disagrees with host -t ns digium.com.
14:39.59fileit was a registrar problem, it's slowly resolving itself
14:40.14jkroonok cool.  then i'll just wait.
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16:24.52jsmithwaves
16:25.04jsmithIt's been a while since I've hung out in here, so I'll say hi :-)
16:27.24seanbrightwell if it isn't the VP of... community something
16:27.50seanbrightor "mr. fancy pants" as i like to call you
16:27.53filewarbles
16:28.12seanbrighti was in my hotel room the night of the all conference happy hour and was wondering if it was still going on
16:28.33seanbrightso i glanced out the window and saw the bright orange pants and knew things were still in full swing
16:34.06SamotOh please tell me is was file in bright orange pants and a white top hat.
16:34.15SamotBecause, that would be brave.
16:34.17seanbrightno, it was jsmith
16:34.21filejsmith had the pants
16:34.22fileI had the hat
16:34.38fileapparently I was known by some as "the guy with the hat"
16:34.51seanbrighti had a white carnation in my lapel. it was all very romantic.
16:35.18SamotBetter than the guy with the pants.
16:35.40Samot"Which guy?"
16:36.04Samot"They guy wearing the pants that make you question is he a road worker or an escaped con"
16:36.54SamotDid said pants have reflective panels?
16:37.14SamotI mean were they "night time" safe?
16:38.30sibiriaor did they make him look like the local county jail was a cowboy short?
16:44.49jsmithSamot: Sorry, no reflective panels...
16:45.33SamotI think you missed out on a serious statement.
16:45.49SamotI'm not sure what that statement would have been. But those pants would have made it serious.
16:46.52SamotNaw, I'm kidding. No one takes anyone wearing orange pants casually as serious.
16:47.24seanbrighti have no idea what is going on
16:47.33seanbrightjsmith was wearing bright orange pants. end of story.
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16:49.35SamotHey, I used to wear Hawaiian shirts to the office because the dress code said shirts had to have a collar.
17:00.04igcewielingI 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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18:25.39bkervaskiHey 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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18:30.10fileyou'd need to show the SIP request in question
18:30.24filebut it seems to think that the SDP has no m= line
18:31.42bkervaskiIt doesn't, looking at the request.  Just v,o,s,c,t .. no "m"
18:32.15bkervaskidirect media related?
18:32.45filethat means the SDP has no media streams, we require one
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18:34.11fileunless you mean that we are sending an SDP without a media stream and getting back a 400 Bad Request
18:36.28bkervaskiIt goes out without the m header .. but I think it's UDP related on my development vm .. digging into it
18:36.51bkervaski.. as local sip calls do have =audio 50128 RTP/AVP 9 0 101
18:36.59bkervaskim==audio 50128 RTP/AVP 9 0 101
18:37.23filenods
18:37.55fileso 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
18:38.54bkervaskiCorrect.
18:39.06bkervaski(will do) thanks
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19:47.32Banana51What 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.
19:48.17Banana51I will connect to my asterisk server, which is actually on the same VPS
19:56.24lagzillaI don't know about best but google revealed https://github.com/AGProjects/python-sipsimple
19:57.37Banana51thanks 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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20:47.40bkervaski.. codecs .. wasn't allowing any codecs .. doh!
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21:59.41Banana51? it allows codecs, but send/reads the data stream directly to capture/playback device and not buffers that I can access
22:01.39Banana51edit: never mind
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