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| 05:33.31 | Jason_weileilei | #freeswitch |
| 05:35.33 | ChannelZ | #sogothen |
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| 05:54.42 | drmessano | lol |
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| 06:20.28 | ChannelZ | I don't get that guy. |
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| 08:21.56 | janicez | ChannelZ: He's worse than me, amirite? |
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| 11:19.33 | refj | Is it recoomended to run Asterisk 13.4 on CentOS 7 for production? I have no problems writing systemd unit files if it is needed. |
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| 11:29.18 | hirogen | hi |
| 11:29.58 | hirogen | anyone got a guide on using avay asite adminsitration jsut liek the commands to craeate usesr and swap users and create and change pick up groups etc.. |
| 11:30.01 | hirogen | woudl be handy |
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| 13:09.42 | Ice_Strike | WIMPy You around? |
| 13:09.57 | WIMPy | Here! |
| 13:10.08 | Ice_Strike | Great! May I PM for a sec? |
| 13:10.16 | WIMPy | yes |
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| 13:37.46 | refj | Is there an official reason for not supplying Asterisk 13 LTS repos for CentOS 7 and Ubuntu 14.04 LTS? |
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| 16:15.12 | tompaw | Morning. |
| 16:15.32 | pjensen00 | Is it morning over there? |
| 16:15.45 | tompaw | 11 AM ;) |
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| 16:16.50 | tompaw | pjensen00: I got my "cluster" in place, 3 boxes ready to go, will start experimenting live today, see where it gets me (hopefully not the asylum) |
| 16:17.22 | pjensen00 | Awesome! If you need any ARI help or want to throw ideas off my face let me know! |
| 16:17.31 | tompaw | Will do, cheers. |
| 16:18.50 | tompaw | However, before I get there... a 13.3 box, 3 days uptime, 3 hours of heavy (35%) load (6c w/ht) - out of a sudden asterisk crashes, nothing in syslog. restarting (incl. killing the process and starting it again) spawns zillions of "Exceptionally long voice queue length queuing to ..." errors and no registrations are possible |
| 16:19.01 | tompaw | Only restarting the box helped |
| 16:19.54 | tompaw | Does this sound similar to anyone? There's no errors in any of my logs and I'm kinda baffled as to what could've caused it. |
| 16:21.38 | tompaw | The fact that restarting asterisk process didn't help at all makes me thing it's an OS issue, but of what kind? I'd assume some sort of TCP deadlock... if asterisk used TCP. |
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| 16:21.54 | tompaw | ulimit for files is set to 65k |
| 16:22.01 | pjensen00 | I've noticed taht crashes never seem to show up in logs for me. |
| 16:22.14 | pjensen00 | I've killed asterisk a few times and it's usually a bunk dial string |
| 16:22.33 | pjensen00 | something like PJSIP/PJSIP/@ if I really screw things up |
| 16:22.43 | pjensen00 | and then kaboom, all gone. |
| 16:23.37 | pjensen00 | Another one i've done is trying to originate a channel that already exists. Didn't look to close at that one to see if that was the actual crash cause but that.... wasn't good. |
| 16:23.58 | tompaw | but surely that sort of issue wouldn't last through asterisk restart, would it? |
| 16:24.30 | pjensen00 | what do you mean "last through" restart? |
| 16:24.52 | pjensen00 | Oh, and trying to dial an empty string also crashes asterisk. |
| 16:25.04 | tompaw | well, in my case, after it died, I restarted asterisk a couple of times (i.e. manually located the process, killed it, waited, started again) |
| 16:25.19 | tompaw | shit, again. |
| 16:25.48 | pjensen00 | Oh man, I need more coffee. I read everything you wrote again. |
| 16:26.00 | pjensen00 | Clearly my advice has nothing to do with your issue. :( |
| 16:26.53 | pjensen00 | https://issues.asterisk.org/jira/browse/ASTERISK-15360 |
| 16:27.03 | pjensen00 | have you read through any of the asterisk jira issues? |
| 16:27.45 | pjensen00 | I don't think that issue is what you're seeing, but it seems like there's definitely some talk around the error you are seeing |
| 16:29.06 | tompaw | actually, the only thing I added recently was that ChannelLeftBridge event listener... I will try disabling it and see where it gets me |
| 16:32.04 | tompaw | It's horrible cause I don't have that high CPU usage, it simply dies leaving no clues as to what could cause it. |
| 16:32.57 | tompaw | Could it be because I have HT enabled on my CPU? |
| 16:33.06 | tompaw | Few years ago asterisk didn't like it that much. |
| 16:33.06 | pjensen00 | What's your debug/verbosity level for logging? |
| 16:33.09 | file | without a backtrace or full log, there can be only guesses |
| 16:33.16 | tompaw | 4 |
| 16:33.18 | file | Asterisk has never cared about HT |
| 16:33.37 | pjensen00 | Can you post the log? |
| 16:34.12 | tompaw | there was nothing there, the process simply disappeared. let me bump up the file logger to debug |
| 16:34.58 | tompaw | messages => debug,info,warning,error - should that be enough? |
| 16:36.51 | pjensen00 | Possibly. Also, anythign interesting in /var/log/syslog? |
| 16:37.02 | tompaw | actually, I'm gonna route it to my logstash so I can analyze it with kabana |
| 16:37.07 | tompaw | pjensen00: nothing there |
| 16:37.15 | pjensen00 | bummer |
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| 16:38.27 | tompaw | ok, logger reloaded |
| 16:42.49 | tompaw | damn, 0.5 MB/s of logs |
| 16:52.31 | tompaw | ...and now it refuses to crash, obviously <: |
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| 16:55.11 | pjensen00 | You fixed it! |
| 16:56.42 | tompaw | :D |
| 16:57.10 | tompaw | maybe it needed a blow off, like a turbo engine |
| 16:57.28 | tompaw | and that .5MB/s of gibberish is exactly that :-) |
| 16:58.00 | pjensen00 | Mmm.... well whatever it is will surface again most likely. When it does, post the logs of what you have and I can take a look to see if I can find something. |
| 16:58.55 | tompaw | file: are you saying asterisk doesn't benefit from HT in any way? |
| 16:59.07 | file | it may, but it is not aware and doesn't care |
| 16:59.26 | tompaw | I see |
| 16:59.41 | tompaw | pjensen00: will do, turned the event listener back on, we'll see what happens. |
| 17:00.09 | tompaw | but one way or another, the moral of this story is: 10 $900 servers are better than 1 $19k one |
| 17:01.09 | tompaw | ah, crap |
| 17:01.13 | tompaw | lunch break for the agents |
| 17:01.17 | tompaw | there goes my log. |
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| 17:23.48 | jgowdy | My customers are reporting erroneous fax detection on voice calls, are there settings to increase the energy / duration / certainty of fax detection CED tone? |
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| 19:21.47 | marceloamorim | guys, could I use 'hint' in extension context like , exten => _XXX,1,Verbose? |
| 19:22.45 | marceloamorim | I set lots of stuffs before the app dial later on that context |
| 19:24.17 | newtonr | marceloamorim, I don't understand the question |
| 19:24.35 | tompaw | I believe you're asking for NoOp? |
| 19:24.44 | newtonr | are you asking if you can use a pattern match with hints ? like exten => _XXX,hint,PJSIP/BLAH ? |
| 19:26.18 | marceloamorim | I use a general context for my extensions, like I put the extensions in a database, so I can set this general context to use to call everyone without specified everytime |
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| 19:29.45 | marceloamorim | but every exemple I saw for hint, the context specified just the hint on it, exten _111,hint,SIP/person1&SIP/person2, my goal is set exten => _111,hint,Dial(SIP/ODBC_PERSON,,) after an exten => _111,n,Set(extens_x=${ODBC_blabla}) |
| 19:30.03 | tompaw | ok, so |
| 19:30.12 | tompaw | My asterisk just crashed again |
| 19:30.17 | tompaw | with debug on |
| 19:30.20 | tompaw | nothing in logs :( |
| 19:31.00 | overyander | I'm trying to use blink softphone to establish a video call between two laptops. in my asterisk debug logs i found the following and think it might be the issue. BTW, the video codec in blink is set for h.264 and i confirmed that codec is on my system (core show codecs) Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) |
| 19:31.09 | overyander | why does it show the video codec there as "nothing". I have videosupport=yes and textsupprt=yes set for each of the peers involved in the call |
| 19:33.01 | marceloamorim | tompaw: its crashed after some action, or without no reason? |
| 19:33.50 | tompaw | marceloamorim: no reason, no logs, nothing in syslog / messages / kern / asterisk-debug |
| 19:33.58 | tompaw | it just *poofs* |
| 19:34.22 | marceloamorim | remove all modules and use just the essential |
| 19:35.00 | [TK]D-Fender | overyander, where did you tell it which CODECS it could use? |
| 19:35.33 | marceloamorim | once I had to remove everything to see that my mysql was the problem =( |
| 19:35.55 | overyander | i was just looking at that myself. i noticed that in my allow it was only set for ulaw. I'm chaning that be ulaw,t140,h264 so as to allow audio, text and video. |
| 19:36.29 | marceloamorim | but I'm newbie "dev/user", maybe someone else should help you =) |
| 19:37.02 | tompaw | marceloamorim: I cannot "remove everything" because without res_postgres and ari it's useless to me. |
| 19:38.24 | tompaw | I can try upgrading to 13.4 (on 13.3.2 atm) |
| 19:39.25 | tompaw | were there any serious stability issues with 13.3.2? |
| 19:39.45 | marceloamorim | but if you remove those 2 modules and the asterisk don't crash, you have the road to follow |
| 19:40.03 | tompaw | if I remove those 2 modules I am 100% sure it will not crash because not a single agent will log in. |
| 19:40.09 | tompaw | yay! |
| 19:40.39 | tompaw | it's like healing pneumonia with "try not breathing" |
| 19:42.03 | [TK]D-Fender | marceloamorim, "_111,hint,Dial(SIP/ODBC_PERSON,,)" this is NOT a thing |
| 19:42.23 | [TK]D-Fender | You do not put APPLICATION on a "hint" priority. You seem to not understand what a "hint" actually is |
| 19:43.50 | tompaw | one thing I can try disabling is g729 |
| 19:49.54 | tompaw | This is killing me :-( |
| 19:53.38 | tompaw | why doesn't it produce any sort of trace when it crashes |
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| 20:02.24 | marceloamorim | [TK]D-Fender: http://pastebin.com/VhGYGV4E I use this one |
| 20:04.26 | tompaw | it's not g729, it's not ari events |
| 20:05.18 | tompaw | the logs are useless, what else can I do? |
| 20:05.57 | overyander | [TK]D-Fender, now i'm getting the error "Ignoring video stream offer because port number is zero" |
| 20:06.09 | overyander | i have audio, but no video. |
| 20:10.01 | tompaw | "Failure to build Asterisk with shared pjproject object libraries WILL result in seemingly random crashes. For Asterisk to work properly with pjproject, pjproject MUST be built with shared object libraries." |
| 20:10.12 | tompaw | I found this in your docs |
| 20:10.30 | tompaw | Could it be the reason? |
| 20:10.40 | tompaw | Or would PJSIP not work at all if it wasn't properly linked? |
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| 20:54.21 | marceloamorim | o.O Just need to set callcounter=yes on sip.conf and the hint is already set properly |
| 20:54.37 | marceloamorim | without you need to put on extensions.conf |
| 20:56.29 | [TK]D-Fender | .... |
| 20:56.45 | [TK]D-Fender | setting that on a peer is not the same as putting ahint in the dialplan at all |
| 20:57.53 | marceloamorim | when you set this, in dialplan show hints, the asterisk can set inuse properly, without this callcounter the asterisk didn't set |
| 20:58.35 | marceloamorim | there is other ways to do that, but this way is fine for what I want, I didn't know |
| 20:58.44 | marceloamorim | probably* |
| 21:01.01 | [TK]D-Fender | You need that for hint's to properly track a status. But that has nothing to dow ith not NEEDING "hints" |
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