| 00:13.50 | [J]oules | file: are you a developer for digium / asterisk? |
| 00:13.55 | file | yes. |
| 00:14.01 | [J]oules | wow, cool |
| 00:19.51 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:19.51 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:29.34 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
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| 01:09.21 | Samot | file: is the shiznit |
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| 01:55.06 | rpifan | hello |
| 03:18.19 | *** join/#asterisk ledoktre (~Adium@216.51.224.229) |
| 03:19.34 | ledoktre | greets. im having problems with call parking. I have the feature enabled, but try as I might it sees like I cannot get the call to park into the specified parking lot. Always goes to default lot. I set the ${CHANNEL(parkinglot)} variable before the call, but it still does not go. Help? |
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| 06:39.26 | jkroon | does anybody know if the security framework also supports successful registrations (authentications)? |
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| 12:43.59 | *** join/#asterisk jchillerup (~jens@host-185-57-161-120.adsl.gl) |
| 12:46.19 | jchillerup | Hi. I'm trying to set up a system in which I have telephones inside of a VPN IP range (10.0.0.0/8) that need to able to call out through our SIP trunk. Asterisk runs on the same server as the VPN, so it can see both the internet and the innards of our VPN. |
| 12:46.50 | jchillerup | Now, I can place a call by having my transport bind to 0.0.0.0 but then voice data is lacking for calls going out via the SIP trunk, presumably because the source IP in the SIP packets is wrong. |
| 12:49.32 | Samot | Do you have your local networks and external IP details setup correctly in Asterisk? |
| 12:50.31 | jchillerup | Good question, let me make a pastebin |
| 12:51.15 | jchillerup | https://pastebin.com/42zcg8As |
| 12:51.33 | jchillerup | Maybe what I should do is to make a separate transport for each interface? |
| 12:51.43 | jchillerup | Thing is, I also want my SIP clients to be able to connect outside of the VPN |
| 12:51.46 | Samot | Each interface? |
| 12:52.04 | jchillerup | VPN interface and the "internet interface" |
| 12:52.22 | jchillerup | VPN = 10.0.0.0/8, internet = 130.225.212.254 |
| 12:52.48 | Samot | Well first you should show a call. |
| 12:52.52 | Samot | asterisk -rvvvvvvvvvv |
| 12:52.56 | Samot | pjsip set logger on |
| 12:53.09 | Samot | Make a call and show the results. |
| 12:53.19 | Samot | Let's see how an external call actually looks. |
| 12:54.07 | jchillerup | hmm, that was too many lines for my terminal app |
| 12:54.10 | jchillerup | hang on... |
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| 12:57.14 | jchillerup | Samot: https://pastebin.com/AQNF8ce9, action starts at line 37 |
| 12:58.21 | Samot | c=IN IP4 192.168.52.2 |
| 12:58.44 | Samot | From: "FOM" <sip:301@192.168.52.2>;tag=65e88018-b919-4797-a43f-6fd4f70b7a37 |
| 12:58.55 | Samot | Where is the 192.168.52.x network coming from? |
| 12:59.06 | jchillerup | whoops |
| 12:59.13 | jchillerup | remnants of an old DNAT setting |
| 12:59.52 | Samot | Well that is the IP Asterisk is using for this. |
| 13:01.09 | jchillerup | It should be 130.225.212.254 |
| 13:01.29 | jchillerup | So, that would be solved by setting the appropriate external_signalling_address, right? |
| 13:02.29 | [TK]D-Fender | <--- Transmitting SIP request (933 bytes) to UDP:213.83.164.141:5060 ---> |
| 13:02.46 | [TK]D-Fender | Contact: <sip:asterisk@130.225.212.254:5060> |
| 13:02.54 | [TK]D-Fender | Seems to by using the right IP in the contact |
| 13:02.57 | [TK]D-Fender | #164 |
| 13:03.16 | [TK]D-Fender | c=IN IP4 192.168.52.2 <- #179 hrm |
| 13:04.09 | jchillerup | maybe a quickfix could be to set local_net=192.168.0.0/16 . I'm not using that range for any endpoints anyway |
| 13:04.40 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:04.46 | file | you should also not be using certified 13.13, because I do recall fixes in the NAT area |
| 13:05.09 | jchillerup | Hmm.... |
| 13:07.27 | jchillerup | Funny thing is that it used to work. I just don't recall what's changed (and I can't call the other sysadm ;)) |
| 13:07.44 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-rqsxrywxqpwlvuuh) |
| 13:10.31 | Samot | Well you shouldn't be using Certified unless you have a support deal with Digium. |
| 13:10.36 | Samot | Otherwise, it's really pointless. |
| 13:11.26 | Samot | You get absolutely no benefit from the Cert version AND you put yourself in a position where you are behind on current updates. |
| 13:12.43 | jchillerup | That's good to know |
| 13:12.53 | jchillerup | Guess I should revise that. |
| 13:13.04 | Samot | I think the latest Cert version is 13.18 |
| 13:13.17 | Samot | And the current version release is 13.20 |
| 13:13.25 | *** join/#asterisk ryang (sid10904@gateway/web/irccloud.com/x-zrrwxdtvnoguftos) |
| 13:13.29 | jchillerup | I picked it becuase it had a 'this one is the stable one' feel to it |
| 13:13.37 | jchillerup | Also because I don't need anything fancy |
| 13:13.41 | Samot | They are both stable. |
| 13:13.47 | Samot | Cert gets less updates |
| 13:13.49 | Samot | Well |
| 13:13.54 | Samot | Less update roll outs. |
| 13:14.00 | jchillerup | ya, makes sense. |
| 13:14.07 | jchillerup | Anyway, call-IN works, also with sound |
| 13:14.08 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-uvmgoksvvxigqgpg) |
| 13:14.45 | jchillerup | So that source IP 192.168.52.2 is probably the culprit. Dropped a mail to the other sysadmin who's on-site to have a look. |
| 13:14.56 | Samot | So when the next Cert version is released it will have all the updates between 13.18 and that version. |
| 13:15.03 | Samot | You just have to wait longer for fixes. |
| 13:15.23 | jchillerup | Makes sense. However, I don't think my problems here relate to a software bug. |
| 13:21.18 | jchillerup | Thanks for the help, Samot |
| 13:22.56 | *** join/#asterisk rwb (~Thunderbi@74.85.159.242) |
| 13:26.32 | sibiria | marking the LAN with local_net does nothing? |
| 13:31.38 | jchillerup | No, not by the looks of it |
| 13:31.49 | jchillerup | Symptom is the same anyway, haven't checked the logs |
| 13:34.47 | sibiria | i think we solved a similar case by local_net and externhost |
| 13:35.01 | sibiria | or was that a binat setup that was giving problems... can't recall |
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| 15:05.32 | igcewieling | Verizon SIP has an outage, giving 606 replies when it shouldn't: "Incident Isolated --- The issue has been correlated to a major outage impacting multiple customers. We will provide updates as soon as possible. |
| 15:05.48 | jkroon | res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '"'or''='" <sip:'or''='@154.73.34.17>' failed for '107.155.186.103:5076' (callid: 98f6c5ac89c8f0a5dd7a5c9c35dc5400) - No matching endpoint found <-- SQL injection? seriously? |
| 15:09.49 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 15:20.44 | Samot | How is that an SQL injection? |
| 15:20.52 | Samot | Looks like an INVITE. |
| 15:25.02 | jchillerup | '"'or''='" |
| 15:25.08 | jchillerup | That looks kindda fishy to me ;) |
| 15:37.09 | *** join/#asterisk kharwell (kharwell@nat/digium/x-feskygfuzrhfqbcz) |
| 15:37.09 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:37.51 | Samot | Yes it does. But it's a SIP INVITE. |
| 15:39.40 | Samot | The only way that becomes a SQL injection is if the SIP server is actually writing rejected/failed requests to SQL *AND* the data is not being sanitized before insertion. |
| 15:40.25 | Samot | If you're letting public/unknown sources make insertions into the database, you're dumb for not doing any sanitizing to that data. |
| 15:47.32 | jchillerup | yeah yeah |
| 15:47.55 | jchillerup | I dunno, some folks might log these things in a db? |
| 15:53.32 | jchillerup | Anyway, I'm off. Thanks for the help earier, Samot! It's definitely related to the source IP being wrong, so I'm going to investigate! |
| 15:53.58 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 16:21.04 | *** join/#asterisk blu_ (~bluenemo@unaffiliated/bluenemo) |
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| 16:25.37 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 16:30.38 | Worldexe | do we have some place in dialplan to be called exactly once, at asterisk startup (on fake channel, maybe)? |
| 16:31.04 | Worldexe | i need a couple cleanup queries to be executed |
| 16:31.15 | Worldexe | well, i can always hack init script... |
| 16:31.58 | igcewieling | you'll have to hack the init script |
| 16:32.37 | Worldexe | or i can hack asterisk... |
| 16:32.48 | igcewieling | you could always back up some sort of asterisk -rx "originate blah blah blah" to force a call. |
| 16:33.25 | igcewieling | Worldexe: you mean you can hack Asterisk and then spend time integrating your patches into any new Asterisk versions you install? |
| 16:34.09 | [TK]D-Fender | What are you actually trying to accomplish? |
| 16:34.44 | Worldexe | well, yeah, patch-management is not a thing i want to spend time on... |
| 16:35.36 | [TK]D-Fender | Does it have to be as of the moemnt * actually starts and ever time it does (like if * rashes and the init script restarts it, etc) |
| 16:35.44 | Worldexe | i have current calls state being written to db (via func_odbc); in case Asterisk crashed or something went wrong, those tables should be cleaned up |
| 16:36.41 | [TK]D-Fender | Sounds liek init script mod is the way to go. |
| 16:37.13 | file | there's a thing... |
| 16:37.40 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/cli.conf.sample |
| 16:37.51 | Worldexe | ouch! |
| 16:37.54 | Worldexe | thanks! |
| 16:38.06 | file | which could do as igcewieling stated, an originate... to execute dialplan using a Local channel... |
| 16:38.17 | Worldexe | yeah-yeah, i got it |
| 16:44.26 | Worldexe | file, i have a couple patches i wrote (adding features like being able to stop Monitor on another channel and similar things); can i just post them to bugtracker? is it the way to suggest such things? |
| 16:44.56 | file | the contribution process is on the wiki, https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process |
| 16:45.00 | Worldexe | thx |
| 17:05.25 | *** join/#asterisk Oooohboy (~Oooohboy@cpe-67-11-10-120.satx.res.rr.com) |
| 17:05.37 | *** join/#asterisk dar123 (~dar@2600:1700:38d0:1470:7057:bae4:5d80:3cab) |
| 17:26.37 | *** join/#asterisk akay_ (akay@unaffiliated/akay) |
| 17:42.22 | [J]oules | file hi, you have any idea how to find the cause of: res_pjsip/pjsip_distributor.c:538 distributor: Taskprocessor overload alert: Ignoring 'Request msg REGISTER/cseq=1 (rdata0x7f58bc8b21d0)'. |
| 17:42.38 | [J]oules | our pbx drops all extensions |
| 17:42.50 | file | I already answered that in #freepbx when you originally brought it up over potential reasons |
| 17:42.53 | file | and you've filed an issue already |
| 17:43.07 | [J]oules | oh yeah. i did you're right |
| 17:44.17 | [J]oules | shows 187 taskprocesses but that doesn't mean much i presume |
| 17:44.36 | file | rmudgett already responded on your issue asking for info |
| 17:45.19 | [J]oules | he did? |
| 17:45.24 | [J]oules | ok i go check email |
| 17:47.05 | [TK]D-Fender | #waybackmachine |
| 17:58.19 | [J]oules | file: nothing in email from asterisk or digium |
| 17:58.37 | file | https://issues.asterisk.org/jira/browse/ASTERISK-27821 |
| 17:59.17 | file | it also did generate email, a copy went to http://lists.digium.com/pipermail/asterisk-bugs/2018-April/185357.html |
| 18:19.09 | *** join/#asterisk twanny796 (c39e6f10@gateway/web/freenode/ip.195.158.111.16) |
| 18:30.47 | *** part/#asterisk kharwell (kharwell@nat/digium/x-feskygfuzrhfqbcz) |
| 18:42.18 | [J]oules | file: the mail comes from asterisk.org? |
| 18:42.45 | file | that would be the From address |
| 18:43.00 | [J]oules | nothing hit our mail server from asterisk.org |
| 18:44.35 | [J]oules | H=mail.digium.com [216.207.245.2]:51586 X=TLSv1.2:ECDHE-RSA-AES128-GCM-SHA256:128 CV=no F=<noreply@issues.asterisk.org> temporarily rejected RCPT Could not complete sender verify callout |
| 18:45.02 | [J]oules | i will add digium address so it can get through |
| 18:45.15 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-goaobfddhqidgrrp) |
| 18:45.15 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 18:52.21 | *** join/#asterisk blu_ (~bluenemo@unaffiliated/bluenemo) |
| 19:03.27 | igcewieling | I just saw one of our sales reps ordering a "non-digital" Polycom sidecar. *sigh* |
| 19:05.00 | Samot | That should be interesting. |
| 19:05.27 | Samot | Did you tell him he has two options, B/W or Color? |
| 19:05.46 | igcewieling | No. It won't. It will be annoying, harsh words will be exchanged and the customer will be unhappy. |
| 19:06.08 | Samot | Well it is hard to order something that doesn't exist. |
| 19:06.08 | igcewieling | For the most part our sales rep ignore my e-mails so I don't bother anymore. |
| 19:06.24 | Samot | Distributors just can't fill the orders. |
| 19:06.54 | igcewieling | Eventually someone will guess and order a bw sidecar or everyone will ignore the invalid order until the customer complains. |
| 19:21.16 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 19:42.53 | qakhan | hi all, is there any way we can save Manage events in log file on the Astersik server |
| 19:47.28 | [TK]D-Fender | tcpdump |
| 19:47.44 | [TK]D-Fender | Nothing in * unless you have core debug all the way up and that will get you a lot more junk |
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| 20:05.04 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 20:05.21 | [J]oules | rmudgett: got your mail about symbols. |
| 20:07.36 | [J]oules | also your comments about moh, well, we have other pbx's which get a lot of calls but they don't use pjsip and never loose registration. The moh is radio station broadcast and the other pbx's are unaffected by it. Only our pbx is doing this. |
| 20:11.46 | *** join/#asterisk Sepultura (~Sepultura@unaffiliated/sepultura) |
| 20:22.30 | rmudgett | Which is why I want a full backtrace to see what those threads are blocked waiting for. |
| 20:54.51 | errentazaria | Is it possible to send NOTIFYs to only 1 contact in PJSIP? |
| 20:56.14 | [J]oules | rmudgett: hi, did a search for installing symbols freepbx but didn;t find any guide or how-to |
| 20:57.02 | [J]oules | rmudgett: this issue is only affecting pjsip ext's. |
| 21:01.25 | file | he linked you to a wiki page on the FreePBX wiki with details |
| 21:03.02 | [J]oules | talking about sudo /var/lib/asterisk/scripts/ast_coredumper --running ? |
| 21:06.37 | Samot | Here's an idea.. |
| 21:06.45 | Samot | Turn off the moh method that you are using |
| 21:06.49 | Samot | Does the issue go away? |
| 21:07.21 | rmudgett | That will get the backtrace but if the symbols aren't install then there won't be any. I know there is a freepbx page documenting how to install symbols but I don't know where it is. |
| 21:08.25 | [J]oules | rmudgett: like I mentioned, search for 'install symbols freepbx asterisk' shows everything but what you want |
| 21:09.27 | rmudgett | Yeah, I tried that when I was creating the response request. You could ask on #freepbx where it is. |
| 21:10.15 | Samot | https://wiki.freepbx.org/display/SUP/Providing+Great+Debug |
| 21:10.19 | [J]oules | Samot: this pjsip problem occurs at any time. At ~ 11pm, our radio server shuts down to preserve bandwidth. And at that point see non stop warning messages because moh cant get anything. But this happens to all pbx's/. Only our pbx extensions drop off and cant reg |
| 21:10.19 | file | it's at the bottom of the wiki page. |
| 21:10.27 | Samot | If you're running FreePBX Distro 7, you can simply install the 'sangoma-devel' package and then use the new 'debuginfo-install' command to download all the required debug packages. Note that you should replace the asterisk version in the command with the actual asterisk version that you are running ('asterisk11' or 'asterisk14', for example) |
| 21:11.15 | Samot | [J]oules: Do you change the music on hold method? How can you have MoH stream from a server that is shut down? |
| 21:11.31 | [J]oules | Samot: i dont think that link is to install symbols |
| 21:11.48 | Samot | Dude |
| 21:11.49 | [J]oules | Samot: no we dont change anything |
| 21:12.01 | Samot | In order to install "symbols" you have to install devel packages. |
| 21:12.05 | Samot | That's what that is. |
| 21:12.13 | [J]oules | let me look at it again |
| 21:12.23 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:12.51 | Samot | [J]oules: OK, so you stream MoH but then you shut down the server that your stream connects to but you don't change MoH that is being used for calls... |
| 21:13.01 | Samot | So what happens if there is a call that requires MoH? |
| 21:13.04 | Samot | Dead stream?! |
| 21:13.08 | [J]oules | yes |
| 21:13.14 | Samot | OK |
| 21:13.19 | Samot | So you have MoH trying to play |
| 21:13.20 | [J]oules | since no one is using phones between 11pm and 8 am |
| 21:13.22 | Samot | But then timing out |
| 21:13.31 | [J]oules | no... i will show you.. once sec |
| 21:14.09 | [J]oules | WARNING[1487] res_musiconhold.c: poll() failed: Interrupted system call |
| 21:14.24 | [J]oules | millions of those every night |
| 21:14.30 | Samot | YES |
| 21:14.37 | Samot | Because MoH stream is trying to stream. |
| 21:14.55 | [J]oules | yeah but it doesn't affect any other pbx. and our pbx is hte ONLY one using pjsip |
| 21:15.00 | Samot | OK |
| 21:15.06 | Samot | So then it's an issue with PJSI |
| 21:15.12 | [J]oules | exactly |
| 21:15.17 | Samot | So then it's an issue with PJSIP not wanting to waste resources |
| 21:15.25 | Samot | As file has explained. |
| 21:15.39 | [J]oules | i still dont see what to install on that page.. https://wiki.freepbx.org/display/SUP/Providing+Great+Debug |
| 21:15.55 | Samot | # Only for FreePBX Distro 7! |
| 21:15.55 | Samot | yum install -y sangoma-devel |
| 21:15.55 | Samot | debuginfo-install --enablerepo=centos7-debuginfo asterisk14 |
| 21:16.06 | Samot | It is literally right there.. |
| 21:16.17 | Samot | At the bottom of the page. Again, just as file pointed out. |
| 21:16.47 | [J]oules | sangoma-devel' package |
| 21:16.59 | [J]oules | OK |
| 21:17.03 | [J]oules | got it |
| 21:17.07 | [J]oules | thank you |
| 21:17.30 | Samot | This is completely related to your MoH stream. |
| 21:17.45 | Samot | I'm with rmudgett on this. |
| 21:19.07 | [J]oules | so i need yum install pjproject-debuginfo asterisk13-debuginfo |
| 21:19.09 | rmudgett | Heh. I missed that step where they install the devel package when getting the backtraces. I thought it was a specific page. |
| 21:19.17 | [J]oules | since we have asterisk 13.19.1 |
| 21:20.20 | Samot | Dude |
| 21:20.24 | Samot | Are you on SNG7? |
| 21:20.32 | [J]oules | yes |
| 21:20.43 | Samot | Then what I posted is what you need to use |
| 21:20.49 | Samot | Change asterisk14 to asterisk13 |
| 21:20.57 | Samot | The instructions can't get any clearer. |
| 21:21.29 | [TK]D-Fender | That would require reading.... |
| 21:21.51 | Samot | [J]oules: You understand that when you use the application option for MoH it's a *stream* which means it's always streaming. |
| 21:22.08 | Samot | Regardless if there is an active call or not. That's why it's a stream. |
| 21:22.37 | Samot | So when you shut down the server that should be streaming the audio for MoH your stream application is going to have some issues... |
| 21:24.49 | [J]oules | so i guess my only alternative is to leave radio station broadcasting all night |
| 21:24.59 | Samot | 5:13:21 PM <[J]oules> since no one is using phones between 11pm and 8 am <-- So none of the PBXes using the radio server for MoH have calls during those hours? Not a one? They all have hours that start after 8am? |
| 21:25.03 | [J]oules | by the way it only installs like this: yum install -y sangoma-devel debuginfo-install asterisk13 |
| 21:25.20 | Samot | Dude.. |
| 21:25.21 | [J]oules | yes to your question |
| 21:25.26 | Samot | Those are two different lines.. |
| 21:25.31 | Samot | Two different commands. |
| 21:26.03 | [J]oules | ok |
| 21:28.43 | [J]oules | installed both |
| 21:28.55 | [J]oules | but the rest of the page implies core dump |
| 21:29.20 | [J]oules | so what would I run to get the log rmudgett wants |
| 21:30.09 | file | he linked to a wiki page which has instructions. |
| 21:31.18 | rmudgett | You can use ast_coredumper that is described there or you can manually use gdb to get the backtrace from a running instance. |
| 21:33.38 | [J]oules | running /var/lib/asterisk/scripts/ast_coredumper /tmp/file-name is for a core dump, we dont have any coredumps |
| 21:34.08 | rmudgett | Manually the command would look someting like: gdb -ex "thread apply all bt full" --batch /usr/sbin/asterisk `pidof asterisk` > /tmp/backtrace-threads.txt |
| 21:34.20 | rmudgett | ast_coredumper --running |
| 21:34.26 | file | our wiki page includes for both a core dump and a deadlock, you want for deadlock |
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| 21:37.40 | [J]oules | ok i ran it like you wrote rmudgett |
| 21:37.47 | igcewieling | Wow, I just realized it is Friday |
| 21:38.06 | [J]oules | but i have to wait till the issue reoccurs |
| 21:38.11 | [J]oules | its is igcewieling |
| 21:38.21 | [J]oules | igcewieling: too much partying? |
| 21:38.42 | rmudgett | igcewieling: Ask file what day it is. He seems to think it is the weekend. :) |
| 21:39.17 | igcewieling | [J]oules: no, I usually don't pay attention to what day it is. I've been busy all day. |
| 21:40.08 | file | rmudgett: it is! not my fault you work later hours |
| 21:41.16 | [TK]D-Fender | weekend !=WORKend |
| 21:41.27 | [J]oules | are these going to be problematic with backtrace: |
| 21:41.34 | [J]oules | warning: Could not load shared library symbols for /usr/lib64/asterisk/modules/app_flite.so. |
| 21:41.34 | [J]oules | Do you need "set solib-search-path" or "set sysroot"? |
| 21:41.40 | file | [TK]D-Fender: Potato potato |
| 21:41.45 | [J]oules | warning: the debug information found in "/usr/lib/debug//lib/libpri.so.1.4.debug" does not match "/lib/libpri.so.1.4" (CRC mismatch). |
| 21:41.55 | [J]oules | warning: the debug information found in "/usr/lib/debug/usr/lib/libpri.so.1.4.debug" does not match "/lib/libpri.so.1.4" (CRC mismatch). |
| 21:42.15 | [TK]D-Fender | file, First word there has an obvious capital "P". See, not the same.... |
| 21:44.09 | [J]oules | with that logic, calling the Police is not the same as calling the police. I'll have to remember that |
| 21:46.07 | igcewieling | It isn't. How would the Police who broke up in 2008 help? |
| 21:46.13 | igcewieling | 8-| |
| 21:47.44 | [J]oules | rmudgett: actually looking in /tmp see some core.name.. files would they help? |
| 21:49.08 | rmudgett | Those would be for some other crash. Not necessarily related to the problem you are having. |
| 21:49.25 | [J]oules | ran /var/lib/asterisk/scripts/ast_coredumper /tmp/core.file-name-date.. and it produced a few files |
| 21:50.25 | [J]oules | https://www.irccloud.com/pastebin/yXIytcRK/ |
| 21:51.17 | rmudgett | I'm primarily interested in the full backtrace file. That pastebin is for the locks file. |
| 21:52.35 | rmudgett | The locks file won't have anything interesting in it unless Asterisk is compiled with DEBUG_THREADS |
| 21:54.31 | [J]oules | how about http://pastebin.ca/4017312 |
| 21:57.26 | [J]oules | rmudgett: is that pastebin what you want? |
| 21:58.11 | rmudgett | That's the right file. But it only has function names w/o local variables or line numbers. It also says "No symbol table info available" all over the place. |
| 21:58.47 | [J]oules | hmm |
| 21:59.48 | [J]oules | would that be due to the fact the core file is from yesterday and we only installed the symbols today? |
| 22:01.16 | [J]oules | rmudgett: here is something else, we have backup freepbx, I just checked it and it has the same MOH url and the problematic pbx. the sip phones did not unregister from the backup server. |
| 22:01.50 | [J]oules | s/and the/as the |
| 22:03.11 | [J]oules | i take it there should not be a /tmp/core.filename file if everything is running smoothly |
| 22:03.40 | [J]oules | and if that is true, then the backup pbx has same issue |
| 22:03.49 | [J]oules | because it has a core. file in /tmp |
| 22:04.39 | rmudgett | Core files get generated when a crash happens and you have told it to generate the core file with the -g option. |
| 22:04.53 | rmudgett | The issue you have reported is not a crash. |
| 22:05.29 | [J]oules | there was no crash that I know of. |
| 22:06.19 | [J]oules | but you're saying those core. files only get generated with a crash. Are you talking about asterisk crash, or something causing server to reboot? |
| 22:06.44 | rmudgett | I'm fairly certain that freepbx is setup to restart asterisk automatically if it crashes. |
| 22:07.15 | [J]oules | ok |
| 22:08.52 | [J]oules | so for experiment, I have disabled the entries in crontab to start/stop radio server, so it will remain on all night and day tomorrow. Tomorrow night I will shut down radio server and see what happens on sunday. I know which file you want, so when it occurs again, I will upload it to the ticket/issue |
| 22:09.33 | [J]oules | thanks for your help and Samot's help |
| 22:14.03 | rmudgett | [J]oules: Do look at the generated file to see if there is information in it. Each thread in the file should give the source file name, function name, and line number. https://issues.asterisk.org/jira/secure/attachment/57154/core.174244-thread1.txt is an example of a thread with the symbol information. |
| 22:15.40 | [J]oules | rmudgett: just checked a few pbx servers and they dont have any /tmp/core. files. Only our pbx's and i still think its very much related to pjsip. But i am no coder. |
| 22:17.12 | [J]oules | rmudgett: dont follow what you mean about https://issues.asterisk.org/jira/secure/attachment/57154/ |
| 22:20.54 | rmudgett | That file is showing thread 1's stack. Frame 5 is the clone function in /usr/lib64/libc.so.6. It calls the function in frame 4 which is start_thread in pthread_create.c at line 334 and so on down to frame 0 which is executing pthread_cond_wait in a system library file. |
| 22:21.18 | rmudgett | Each thread in the full backtrace file should have that kind of information detail in it. |
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| 22:41.04 | [J]oules | this: start_thread (arg=0x7f09a97a9700) at pthread_create.c:334 ? |
| 22:47.23 | rmudgett | Frame 5: "#5 0x00007f09a61fd5fd in clone () at /usr/lib64/libc.so.6" calls start_thread which is frame 4 |
| 22:47.58 | rmudgett | Frame 4: "#4 0x00007f09a6ec161a in start_thread (arg=0x7f09a97a9700) at pthread_create.c:334" is at line 334 and calls dummy_start which is frame 3 |
| 22:48.19 | rmudgett | and so on down to frame 0 where the thread is currently executing. |
| 22:50.33 | rmudgett | or in this case blocked waiting for a condition to be signaled. |
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| 23:23.15 | sibiria | this has probably been asked before, but what files generated by menuselect can be copied from one build to the other in order to get the same configuration when compiling? |
| 23:23.21 | *** join/#asterisk sibyakin (~sibyakin@188.162.228.0) |
| 23:23.58 | [TK]D-Fender | https://www.google.ca/search?q=asterisk+menuselect+makeopts |
| 23:26.15 | sibiria | yes the usual advice is that it's just menuselect.makeopts but i've run into differently configured builds despite copying just that file across |
| 23:27.22 | sibiria | there's menuselect.makedeps, and one or two things changed in menuselect/ too - are those the cause? |
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