| 02:18.35 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 02:18.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 02:20.26 | meek424 | yo. anyone has a list of blacklisted numbers that i can import on asterisk db? |
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| 04:00.26 | igcewieling | no, the numbers change all the time. In fact, often they spoof legit numbers. |
| 04:01.00 | igcewieling | Your best option is to send all calls to an IVR and make the caller press a digit to prove they are human. |
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| 07:06.04 | wearyhacker | Straight in at the deep end. Why is is asterisk doing this at "res_pjsip/pjsip_resolver.c:593 sip_transport_is_available(type + PJSIP_TRANSPORT_IPV6)) {" type is already set to PJSIP_TRANSPORT_UDP6, this line seems to ensure this will not work. Anyone got any ideas? |
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| 13:53.42 | DanFromUK | Is it possible to stop logging temporarily when issuing a 'dialplan reload' via AMI? |
| 13:54.54 | DanFromUK | In v1.3 or something, the file logging followed the console, so I could just issue a 'verbose set 0' before the reload. But that won't work in newer versions |
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| 13:55.46 | Samot | Not that I'm aware of. |
| 13:56.00 | Samot | So you want nothing that will output warnings or errors when you do a reload? |
| 13:56.04 | DanFromUK | It's simply that my full logs get flooded when a dialplan reload occurs, which makes it annoying to look through if I need to check what happened to a call. |
| 13:56.43 | Samot | How often are you reloading the dialplan? |
| 13:57.29 | DanFromUK | @Samot: The dialplan is generated by a server process which is tested before going into production. So in production, I don't need/want to log anything on a 'dialplan reload' that's executed automatically. |
| 13:58.03 | DanFromUK | Dialplan reloads occur throughout the day as clients make changes to their call flows on our portal. |
| 13:58.48 | Samot | I'm not sure how you turn it off. |
| 13:59.05 | Samot | Since it issues a dialplan reload and that's the same as doing it in the console. |
| 13:59.19 | Samot | There's nothing in the console that allows you not to see the output. |
| 14:00.25 | DanFromUK | OK. Just wondered if there was a way to console the file logging. The console no longer appears to control the file logging. Issuing 'core verbose set 0' doesn't seem to affect any file logs set up in logger.conf |
| 14:00.39 | Samot | DanFromTheUK: I just did a dialplan reload on a FreePBX box, the only output I got was warnings about all the include files that don't exist. |
| 14:00.40 | DanFromUK | Control* |
| 14:00.43 | Samot | There was like 20 of them. |
| 14:01.02 | DanFromUK | You must have verbose set to 0 or similar. |
| 14:01.10 | Samot | I do asterisk -r |
| 14:01.14 | Samot | dialplan reload |
| 14:01.24 | Samot | Get about 20 or so warnings. |
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| 14:01.53 | Samot | Pure asterisk box, mainly warnings because I have done a few configs. |
| 14:01.58 | DanFromUK | I have verbose set to 3 so I can capture call flow during normal operation |
| 14:02.01 | Samot | But not 20+ like FreePBX. |
| 14:02.16 | Samot | I have my logging set to 3 by default. |
| 14:02.34 | Samot | Dialplan reload only kicks out warning and errors output when I run it. |
| 14:02.36 | DanFromUK | Dunno. I get a line for every line in the dialplan. |
| 14:02.40 | Samot | At most it's like 20 lines |
| 14:02.54 | Samot | So how many lines are being outputted for you on a dialplan reload? |
| 14:03.23 | DanFromUK | Ah. You probably haven't included NOTICE in your logger.conf file |
| 14:03.59 | Samot | full => debug,error,notice,verbose,warning |
| 14:03.59 | Samot | console => debug,error,notice,verbose,warning |
| 14:04.01 | Samot | That? |
| 14:04.43 | Samot | And actually the FreePBX box does kick out a lot of warnings about non existing include files and extensions/priorities already registered that shouldn't be reloaded again. |
| 14:05.34 | DanFromUK | Hmm. I don't know then. I've always been given an output for each dialplan instruction. I'm not by my PC at the moment to share a few examples. Just had a few minutes spare so I thought I'd see if anyone had feedback on it. |
| 14:05.42 | Samot | Now on the FreePBX box I can make a huge chunk of those warnings go away, I turn off the custom context options. |
| 14:06.45 | DanFromUK | I know that I was told you can no longer control the file logging from the console when I upgraded from the version that shared the same verbose number settings to the version that began separating them out. |
| 14:07.06 | DanFromUK | But wanted to see if anything had changed to allow control of file logging from the console or ami |
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| 14:57.09 | fredcwbr-GO | Howdy, |
| 14:57.10 | fredcwbr-GO | Can someone confirm that res_pjsip does accept #include statements on the pjsip.conf ? |
| 14:57.35 | file | it uses the same configuration parser as everything else |
| 14:57.39 | file | #include works, templates work |
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| 14:58.29 | fredcwbr-GO | Thanks., will have to figure what i'm doing wrong,. |
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| 15:14.40 | igcewieling | reading through the sound files, I don't recall seeing this one before. "Thank you for calling to inquire about our comedy club. It is available for loan to discourage further dissemination of bad puns." |
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| 16:51.09 | igcewieling | Was someone asking about odd CPU spikes in Asterisk? I noticed utilization dropped when I set transcode_via_sln = no in asterisk.conf. |
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| 17:23.31 | velix | Interesting. The audio over the university's trunk broke after 3,5 minutes. After raising timers_min_se=99999999 and timers_sess_expires=99999999 it breaks after 14 Minutes. What the heck? |
| 17:25.10 | igcewieling | turn off session timers then |
| 17:26.05 | igcewieling | screwing with the session timer limits is just asking for problems unless you KNOW for sure you need to change then. |
| 17:26.53 | igcewieling | velix: some day you are going to have to stop messing with the PBX and let people actually use it. |
| 17:27.09 | velix | igcewieling: Did so. That was the one with 3,5 minutes. |
| 17:27.50 | velix | igcewieling: I'm having two providers: the University and the commercial trunk. The commercial works. About 1000 calls without a problem. So they're actually using it ;) |
| 17:28.08 | igcewieling | then set session_timers=no? You set session_timers=yes? (without messing with the other timer related defaults) |
| 17:28.32 | velix | I think, it's timers=no in pjsip ? |
| 17:28.57 | igcewieling | correct timers- |
| 17:29.07 | velix | I tried timers=no and timers=yes, had same behaviour. So it seems to be the opposite PBX overriding the values? |
| 17:29.12 | velix | I'm running tcpdump soon. |
| 17:29.20 | igcewieling | well look at the sip packets and look |
| 17:29.34 | velix | yep |
| 17:30.04 | velix | The audio drops, out on both sides. The connection still is up. Pretty annoying to wait 15 minutes all the time :D |
| 17:30.51 | igcewieling | you know timers has nothing to do with audio, right? |
| 17:31.33 | velix | igcewieling: Actually i'm not quiet sure because all the reference to this problems and chan_sip had to do with session_timers. |
| 17:31.38 | velix | quite* |
| 17:32.06 | velix | So I thought pjsip might behave similar and the messing around boosted it from 3,5 to ~ 14 minutes. |
| 17:34.56 | igcewieling | a problem with session timers will tear down the sip connection, which should also tear down the rtp connection. So if your sip connection stops, it might be session timers. but if the sip connection stays up, but the audio drops, then I doubt the issue is with session timers. |
| 17:35.02 | velix | igcewieling: Actually, the part on my side of the PBX is working rocket solid. No student has problems, even not when behind a nat. |
| 17:35.09 | velix | rock solid* ;) |
| 17:35.30 | velix | But how can audio idle out? |
| 17:35.42 | velix | Actually, it's not idle. I was listening the "Picard" via phone :D |
| 17:35.47 | velix | latency is incredible low |
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| 17:48.51 | velix | Okay, *exactly* 03:28 ... since this is reproduceable, let's do some tcpdumping |
| 17:59.01 | igcewieling | you should be able to see the latency with "pjsip show channelstats" |
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| 18:37.00 | velix | igcewieling: RTT is 0.017 |
| 18:37.12 | velix | all others but count are zero |
| 19:04.11 | velix | igcewieling: Interesting, now this pops up (the other PBX is TCP only): Error in refreshing session timer: Unsupported transport (PJSIP_EUNSUPTRANSPORT) |
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| 19:27.31 | igcewieling | this pops up when timers is set to WHAT? |
| 19:27.52 | igcewieling | Have you considered using UDP |
| 19:29.15 | igcewieling | ah, the crappy univ pbx is TCP only. Is their IP PBX something from Microsoft? |
| 19:29.26 | igcewieling | Microsoft Linc? |
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| 20:04.19 | velix | igcewieling: No, innovaphone behind FortiGate. |
| 20:04.28 | velix | I bet with enabled SIP AG... what was it? |
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| 20:04.49 | velix | igcewieling: It pops up when timers=yes |
| 20:05.17 | velix | I've created dumps, but I'm having a hard time anaylzing them. I can record them from my PBX and my client only of course |
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| 20:30.43 | igcewieling | Does the invite indicate the far end supports timers? |
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| 21:17.13 | velix | yes |
| 21:17.54 | velix | Supported: 100rel,replaces,privacy,timer,from-change,histinfo,answermode,uui |
| 21:18.25 | velix | Allow-Events: reg,dialog,message-summary,presence |
| 21:19.58 | velix | It also sends Session-Expires: 1800 |
| 21:20.20 | velix | Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK |
| 21:30.22 | velix | igcewieling: Bad news. When I connect from a SIP client on Windows or Zoiper from my mobile phone, the timeout after 3,5 minutes does NOT appear. I'm trying another thing. |
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| 21:37.21 | velix | D'oh... it's really a problem on the Asterisk PBX. |
| 21:37.40 | velix | I've done a "channel originale ... Application echo()" from my PBX now. Again no audio after 3,5 minutes. |
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| 21:49.47 | velix | I cannot find any reason in the log |
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| 22:12.05 | velix | Ahh, pjsip finally supports OPTIONS via outbould proxy. nice. |
| 22:13.34 | Ai9zO5AP | hi |
| 22:14.24 | Ai9zO5AP | does asterisk have the possibility to test schema of asterisk ? the default schema is public but i want to change it |
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| 22:41.11 | Ai9zO5AP | velix: do you find that pjsip is better than sip ? |
| 22:41.54 | velix | Ai9zO5AP: Some things are really better, but when you freshly started with chan_sip and then switch to pjsip, it's a complete new world. |
| 22:42.17 | velix | For a normal sip setup (not a broken one, like mine), chan_sip is way enough. |
| 22:42.22 | velix | pjsip's user management is aweful. |
| 22:42.32 | velix | aor, endpoint, auth ... |
| 22:42.38 | velix | identify |
| 22:42.53 | velix | You really need scripting, detauls or the wizard for this |
| 22:43.08 | Ai9zO5AP | velix: understood |
| 22:43.21 | Ai9zO5AP | i am working only with sip i feel good, |
| 22:43.41 | Ai9zO5AP | i just switch to postgresql, I managed now to work with schema |
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| 22:49.08 | drmessano | I wish people would stop claiming 'chan_sip is enough' |
| 22:49.13 | drmessano | It's dead |
| 22:49.44 | drmessano | The most significant flaw in chan_sip, is that it's getting nothing new moving forward, and it's on life-support |
| 22:50.16 | drmessano | It's the Windows 7 of channel drivers |
| 22:50.26 | drmessano | It "works fine", but it's completely dead |
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| 22:56.59 | velix | drmessano: Does SIP change that much? |
| 22:57.12 | velix | An OS isn't defined by an RFC. |
| 22:57.17 | velix | SIP etc. are |
| 22:57.26 | velix | They're stable. About no new features. |
| 22:57.34 | file | it does change that much. |
| 22:57.36 | drmessano | New features are added, bugs are fixed |
| 22:57.49 | velix | new ones introduced ;) |
| 22:57.51 | file | and chan_sip was organically written, you can't touch it without something breaking |
| 22:58.12 | drmessano | chan_pjsip actually does modern SIP, unlike chan_sip |
| 22:58.14 | drmessano | Like AORs |
| 22:58.23 | velix | I never said, pjsip is bad. It's just a little bit complicated when coming from chan_ship. |
| 22:58.32 | velix | pjsip has many nice features. |
| 22:58.32 | drmessano | I never said you said it's bad |
| 22:58.39 | drmessano | I said chan_sip is bad |
| 22:58.41 | drmessano | and it's dead |
| 22:59.02 | velix | brb 40 seconds, in 20 seconds, the audio will expire (or not) |
| 22:59.08 | drmessano | and statements like "it's enough" when someone asks "Which is better", should include that chan_sip is dead |
| 22:59.22 | velix | yipeeeeeeeee. it didn't expire! |
| 22:59.29 | velix | I finally broke the 3,5 minutes. |
| 22:59.51 | igcewieling | people who are starting out with Asterisk are idiots if they start with chan_sip instead of pjsip |
| 22:59.58 | drmessano | chan_sip also doesn't work with the modernization some large providers have undergone |
| 23:00.20 | drmessano | Like for example, Flowroute and Twilio are a nightmare with chan_sip |
| 23:00.32 | drmessano | Requiring lots of peers configured for one account |
| 23:01.15 | drmessano | So no, "chan_sip is not enough" |
| 23:01.24 | file | endpoint identifiers are on my list of things I'm proud of, simple but very very useful from an architectual perspective... |
| 23:02.08 | velix | igcewieling: I started with chan_sip because I had a book about Asterisk ;) |
| 23:02.21 | drmessano | If you want an idiot box for your home network, using users.conf is enough |
| 23:03.52 | velix | Anyone with an idea, which this re-registers so often? https://bpaste.net/raw/W3ZA |
| 23:04.56 | file | because the remote side provided an expiration time of 120 seconds |
| 23:04.56 | igcewieling | I see you are still screwing with reg time settings |
| 23:05.47 | velix | file: Yep, just saw it. Expires: 120 |
| 23:06.01 | file | igcewieling: you might find this random information useful - the outbound re-register time in PJSIP is 10 seconds before expiration |
| 23:06.15 | drmessano | But chan_sip is enough, eh |
| 23:06.18 | velix | igcewieling: Yep, the server needs to be hammered. |
| 23:07.02 | velix | It sends a range of unexpected errors sometimes. |
| 23:07.09 | velix | 400, 401, 403 etc. - without a reason. |
| 23:07.14 | igcewieling | file: not at 1/2 of the expiration value? |
| 23:07.19 | file | igcewieling: nope |
| 23:07.33 | igcewieling | file: 10 seconds seems much more reasonable. |
| 23:07.46 | file | agreed |
| 23:08.02 | file | I think I did it, and also that's why I did it |
| 23:10.30 | velix | igcewieling: it really seems to work now (at least for Application Echo). I've simply added `transport = ipv4-tcp` to the endpoint. |
| 23:14.20 | *** join/#asterisk spatel (~spatel@70.42.54.42) |
| 23:32.57 | *** join/#asterisk ShellyRoll (~ShellyRol@c-67-170-30-252.hsd1.wa.comcast.net) |
| 23:35.21 | velix | Sorry for asking, but when does `max_contacts` matter? On call forwarding? |
| 23:36.51 | Samot | It's how many contacts an AOR can have registered to it. |
| 23:37.35 | velix | Okay, I really didn't get behind the AOR concept now :( |
| 23:39.26 | velix | is reading https://wiki.asterisk.org/wiki/display/~jcolp/AORs%2C+Contacts%2C+Device+State |
| 23:39.32 | Samot | Well it's a core SIP concept. |
| 23:39.48 | Samot | Account Of Record. |
| 23:39.55 | Samot | user@domain |
| 23:39.59 | Samot | Basic format. |
| 23:50.44 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@gateway/tor-sasl/ai9zo5ap) |