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| 01:07.53 | mjordan | file: so... did we ever decide on a particular DNS API to use? |
| 01:08.32 | file | mjordan, no |
| 01:08.37 | mjordan | hm. |
| 01:09.59 | mjordan | what's your current leaning? |
| 01:10.08 | file | libunbound |
| 01:11.17 | mjordan | k. |
| 01:11.23 | mjordan | I shalt comment to that effect. |
| 01:11.27 | file | ack |
| 01:23.49 | tm1000 | mjordan: are you still around |
| 01:25.24 | tm1000 | or file if you are free |
| 01:28.25 | tm1000 | boo Well I will just post it up for the hell of it |
| 01:28.39 | tm1000 | I have a bug report that is stating progress() is not working in Asterisk 13 |
| 01:28.42 | tm1000 | dialplan is here: |
| 01:28.43 | tm1000 | http://pastebin.com/tEJn4Bkz |
| 01:29.08 | tm1000 | basically early media, when the user switches back to Asterisk 11, it works |
| 01:34.21 | tm1000 | but when I do it manually it works fine |
| 01:34.26 | tm1000 | I think the user is on something |
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| 01:44.49 | tm1000 | ahhh! |
| 01:45.00 | tm1000 | This is a bug! I can get it to happen with chan_sip but not PJSIP! |
| 01:45.39 | tm1000 | so pjsip is working but chan sip is not |
| 01:45.42 | tm1000 | goes to file bug |
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| 12:26.11 | stefan27 | I don't get it, if I compile with DONT_OPTIMIZE and MALLOC_debug, shouldn't /var/log/asterisk/mmlog be populated with stuff other than 1425037331 - New session |
| 12:26.14 | stefan27 | upon core dumps |
| 12:26.46 | stefan27 | https://wiki.asterisk.org/wiki/display/AST/MALLOC_DEBUG+Compiler+Flag oO? |
| 12:31.22 | wdoekes | stefan27: I thought mmlog is only populated if you tell asterisk to write to it at shutdown |
| 12:31.38 | wdoekes | which it wouldn't do because of the crash |
| 12:33.41 | wdoekes | stefan27: yes, the atexit summary. other than that, it appears to only write (fence/alloc) errors |
| 12:34.10 | wdoekes | (look for astmm_log()) |
| 12:47.24 | stefan27 | thx, that info should be inserted in the wiki |
| 12:50.51 | stefan27 | Also, after latest install with MALLOC_debug, my ast_log(LOG_ERROR, "trying to set dtls conf on %p \n",instance); sometimes outputs short mem-addresses like '0x2f6d558' %p where (instance is struct ast_rtp_instance *instance) all other logs with %p i've ever seen before have been longer like 0x7fa978057200 what governs this |
| 12:51.02 | stefan27 | i thought the address-string had fixed length |
| 12:52.17 | stefan27 | how can i display the virtual address space of a unix process |
| 12:52.18 | wdoekes | I ain't touching the wiki. I just removed the the "s" from "atexist" and it was horrible |
| 12:53.09 | wdoekes | memory address size doesn't change, you're just not seeing the leading 0's |
| 12:54.06 | wdoekes | possibly the "short" addresses are on the stack, but it could be something else. I don't think you should see anything into that |
| 12:54.28 | stefan27 | yeah thats not really related to my issue i was curious though |
| 12:57.06 | stefan27 | pmap <pidof asterisk> -XX shows 1510 memory mapped files with address starting with 7f********************* and 5 weird ones with 00****00 |
| 12:57.35 | wdoekes | (see also /proc/`pidof asterisk`/maps ) |
| 12:58.14 | stefan27 | 0088e000-0089e000 rw-p 0028e000 08:03 531258 /usr/sbin/asterisk |
| 12:58.16 | stefan27 | that's weird |
| 13:03.18 | stefan27 | How come some rtp instances sit inside 00..00 and others in 7f..., where 7fa921109000-7fa921122000 r-xp 00000000 08:03 1185084 /usr/lib64/asterisk/modules/res_rtp_asterisk.so |
| 13:03.56 | stefan27 | I wish I had studied more OS (low level programming) at school |
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| 13:10.43 | stefan27 | shouldn't all ast_rtp_instances get memory addresses matching the range of 012b0000-0398d000 rw-p 00000000 00:00 0 [heap] from cat proc/<pidof asterisk>/maps | grep heap or can they be allocated elsewhere? |
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| 15:18.11 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 13.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev |
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| 15:23.10 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 13.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev |
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| 15:36.53 | stefan27 | newtonr are you the Newton from my jira? I got another coredump with malloc_debug on but log was empty probably due to asterisk being run as asterisk user |
| 15:37.54 | file | he is |
| 15:39.36 | stefan27 | next crash should have a mmlog - but i noticed something weird in the last crash |
| 15:40.01 | newtonr | file, you revealed my secret identity!! |
| 15:44.03 | stefan27 | chan_sip seems to call process_sdp -> process_sdp_a_dtls -> res_asterisk_rtp::dtls_set_setup which ulimately sets SSL_set_connect_state(ssl) OR SSL_set_accept_state(ssl) on both (RTP+RTCP) ssl sessions. But this races with the firing of dtls_perform_handshake(instance, &rtp->dtls, 0); from ast_rtp_on_ice_complete. I'm not sure if this is a problem but in my last crash the ast_ice_on_ice_complete |
| 15:44.03 | stefan27 | fired before dtls_set_setup which i have never noticed during non-crash-calls |
| 15:46.21 | stefan27 | have you been able to get dtls-crashes? |
| 15:49.01 | newtonr | I'll let a dev respond to your query there. I haven't had a chance to try reproducing any dtls crashes in the past weeks no. |
| 15:49.29 | newtonr | If you get a good set of steps on how to reproduce the issue then I can try soon, probably next week |
| 15:49.57 | stefan27 | I need to figure out something smart then |
| 15:51.58 | stefan27 | because it only happens in production with many actual webrtc callers OR with javascript robots executing webrtc calls. |
| 15:52.19 | stefan27 | in super high frequency |
| 15:53.33 | stefan27 | shame there's no "sipp" for webrtc yet |
| 15:53.42 | stefan27 | load tools for webrtc anyone? |
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| 16:01.47 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 13.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev |
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| 16:13.04 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 13.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev |
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| 19:05.53 | stefan27 | why is it always dtls that makes asterisk explode, im using hundreds of asterisks other functions and they never crash |
| 19:07.17 | file | OpenSSL and lack of documentation about it. |
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| 20:57.31 | blitzrage | stefan27, I tend to try and put a proxy in front to do all that and not make asterisk have to deal with it |
| 21:00.50 | pabelanger | ohai |
| 21:01.04 | pabelanger | Did we record the hackathon at astricon this year? |
| 21:02.47 | file | not afaik |
| 21:02.54 | pabelanger | boo |
| 21:03.07 | stefan27 | what kind of proxy might i ask? |
| 21:03.58 | pabelanger | stefan27, we dropped kamailio and rtpengine in front of asterisk to handle webrtc |
| 21:04.10 | pabelanger | in fact our asterisk boxes are still 1.8 |
| 21:04.25 | stefan27 | i used webrtc2sip before as a media gateway from webrtc to asterisk when we had 1.2 asterisk |
| 21:04.26 | pabelanger | took some time to get going, but once running we've been pretty stable so far |
| 21:04.57 | stefan27 | but webrtc2sip gateway had many issues too |
| 21:06.13 | stefan27 | if i can get asterisk 13's dtls stable that'd be preferable, easier to manage |
| 21:11.30 | pabelanger | stefan27, https://www.youtube.com/watch?v=WtqWmmpBSRA |
| 21:11.33 | pabelanger | from astricon this year |
| 21:11.51 | pabelanger | Peter has some good tutorials online how to do it |
| 21:12.13 | pabelanger | I mostly followed his postings about it |
| 21:12.25 | pabelanger | and them some custom stuff for rtpengine |
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| 23:16.02 | pchero | Hello, I'm analyzing Peerstatus message in AMI. But I have a some question. |
| 23:17.10 | pchero | according to this doc,(https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_PeerStatus), The "Cause" section indicate why peerstatus changed. |
| 23:18.07 | pchero | however, when I register it, it doesn't set, when I unregister it set to "Expire" |
| 23:18.33 | pchero | Actually I expected, whatever it is, it's set to correct reason. |
| 23:18.49 | pchero | But, sometimes set, sometimes not. |
| 23:19.01 | pchero | is it bug? or correct? |
| 23:19.55 | pchero | It make me really confusing, I have no Idea how to parsing it. |
| 23:21.21 | WIMPy | Hmm. Is the cause field new? |
| 23:21.38 | pchero | Currently, my program change value only if there's "cause" section in peerstatus. But when peer register it, Asterisk doesn't set any "Cause". It set only when unregister, to "Expire". |
| 23:22.40 | pchero | Actually, not only for "Cause", but also "Time" and "Port" |
| 23:22.50 | WIMPy | Unregister and expire seem to exclude each other. |
| 23:23.12 | pchero | I couldn't see Asterisk set to this value, but I think Asterisk will be set in somehow.. |
| 23:24.23 | pchero | This is what I got when I unregistered.. ["Event":"PeerStatus","ChannelType":"SIP","Privilege":"system,all","PeerStatus":"Unregistered","Peer":"SIP/test-01","Cause":"Expired"] |
| 23:24.57 | pchero | This is for register ["Event":"PeerStatus","ChannelType":"SIP","Privilege":"system,all","PeerStatus":"Registered","Peer":"SIP/test-01","Address":"127.0.0.1"] |
| 23:25.55 | WIMPy | And why is that an issue for you? The cause for a usccessfull registration is obvious. |
| 23:26.31 | pchero | Because I trying to parsing data from AMI... |
| 23:26.48 | WIMPy | That's not a reason. |
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| 23:27.49 | pchero | Well... How can I explain.. |
| 23:28.48 | pchero | At first time, I tried to parsing every section from Peerstatus, and if Asterisk didn't set some section, it update to "NULL" somethig. |
| 23:29.39 | pchero | But..I realized that some information like "Port" are really important. it should not set to NULL.. |
| 23:30.12 | pchero | So, I getting confusing now, what should I do.. |
| 23:32.48 | WIMPy | What do you want? |
| 23:33.07 | WIMPy | As far as I can see the messages contain the fields relecant to the current state. |
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| 23:36.43 | WIMPy | Maybe you should just try to explain your goal. |
| 23:36.59 | pchero | WIMPy: Thank you. :) |
| 23:37.00 | WIMPy | However that might be more appropriate for #asterisk. |
| 23:37.38 | pchero | I create some sqlite3 table for peer's data storing. |
| 23:38.06 | pchero | I wanted that if I get the peerstatus event message, it update table. |
| 23:38.38 | pchero | So.. I tried to use all of section informations.. |
| 23:39.14 | WIMPy | What about only updating the filed you're getting? |
| 23:40.31 | pchero | Yes I did. But problem is.. |
| 23:40.48 | pchero | Currently, the "Cause" is set only "Expired" |
| 23:41.16 | pchero | So.. When I register it again, it still set to "Expire". Because, it doesn't set when it register.. |
| 23:41.33 | pchero | Hm... |
| 23:41.44 | pchero | confusing. :P |
| 23:42.43 | WIMPy | has a feeling that the cause value isn't usefull at all. |
| 23:45.06 | pchero | Ok, I will not use that "Cause. " :P |
| 23:45.12 | pchero | Problem solved! |
| 23:46.40 | WIMPy | That was easy in the end :-) |
| 23:47.38 | *** part/#asterisk-dev mjordan (mjordan@nat/digium/x-dqxnnihmqlzwzcxs) |
| 23:48.07 | WIMPy | However I'd consinder the cause being set to "expired" when someone unregisters to be a bug. |
| 23:48.48 | pchero | WIMPy: Yeap! thanks. :) |
| 23:51.18 | rmudgett | Cause is not always present either. |