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| 01:44.08 | talntid | does anyone know where i can find the table schemas for asterisk realtime for postgres? |
| 01:45.08 | talntid | found some here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL |
| 01:45.19 | talntid | but doesn't seem complete |
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| 01:48.13 | UncleKiwi | hey, can i use templates in the extensions.conf ? |
| 01:57.38 | MaliutaLap | macro? |
| 01:58.40 | UncleKiwi | thanks i'll take a look at macros |
| 02:28.02 | [TK]D-Fender | The closer thing to templates is INCLUDE's, not macros |
| 02:32.21 | carrar | talntid: Did you look in the contrib dir? |
| 02:32.23 | carrar | contrib/realtime/postgresql/postgresql_cdr.sql |
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| 02:33.20 | carrar | or that saem dir for the other scheme's |
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| 07:44.50 | RadJackson | Hi everyone , Is it save to use docker engine to run asterisk ? |
| 07:45.40 | RadJackson | i'm thinking about running two asterisk boxes in the same server using two docker containers, anyone has experienced this ? |
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| 08:32.13 | tzafrir | RadJackson, Why wouldn't it be safe? |
| 08:33.38 | tzafrir | As long as either each instance has their own interface, or you fix binding addresses in the configurations, it would be OK |
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| 08:52.59 | RadJackson | tzafrir each box can receive up to 400 simultenous calls. Docker containers are described as "lightweight" , do you think it can support such load ? |
| 08:53.40 | tzafrir | Docker containers (and other containers) are lightweight as opposed to virtual machines. |
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| 08:54.18 | RadJackson | so you confirm it would be risky to use a container to run an asterisk box with up to 400 simultenous calls ? |
| 08:54.23 | tzafrir | Docker is not the only container technology on Linux. |
| 08:55.06 | tzafrir | Also: why would you run two containers on the same host? If there's a problem with the load, you won't magically get more CPU power |
| 08:56.39 | tzafrir | It's not risky per se. Asterisk may handle more than 400 simulatnious calls (depending on the specific call types and the configurations. Merely "400 calls" means nothing) |
| 08:57.12 | tzafrir | Have you actually tested your typical load? |
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| 08:58.31 | tzafrir | A container is basically a group of processes partitioned in some way from the rest of the system. |
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| 08:58.34 | Tim_Toady | take 'lightweight' with a grain of salt, docker network performance for instance is not that great, far from native and far from what KVM vitrIO can offer |
| 08:59.11 | tzafrir | Tim_Toady, huh? Why is there overhead on docker networking? |
| 08:59.35 | RadJackson | tzafrir here is the use case : we would like to have two asterisk boxes thry containers, one for incoming calls one for outgoing calls , actually we have one asterisk box, and due to the huge amount of simultenous outgoing calls , asterisk CPU gets > 90% , also the memory gets high, so in such case we need to reboot Asterisk , but this has an impact on the current incoming calls. So having |
| 08:59.35 | RadJackson | an independant box only for outgoing calls can let us manage the box without any effect on the incoming calls. And the calls run via IVR, reason for CPU OVERLOAD. |
| 08:59.55 | Tim_Toady | tzafrir: yes, they are using some fancy set up with bridges etc |
| 09:00.30 | Tim_Toady | generally docker networking is still a bit of a mess and that also affects performance |
| 09:00.32 | RadJackson | And Instead of running two asterisk boxes on two separate physical servers, we can handle this case using container on one and unique server. |
| 09:00.44 | tzafrir | RadJackson, this extra separation increases the overhead as there are extra calls between the two boxes, right? |
| 09:01.12 | tzafrir | Why do you need to reboot asterisk? |
| 09:02.21 | RadJackson | Well this is a problem i'm experiencing since months, when my server runs > 450 simultenous calls , CPU gets > 90% and MEM > 10% , even when stopping all outgoing calls , the server keeps lagging and the mem% usage doesnt decrease |
| 09:03.03 | tzafrir | What do you mean by memory usage? |
| 09:03.10 | RadJackson | 450 calls , running through IVR with several mysql connections and AGI scripts calls. The call process is a bit of heavy |
| 09:03.11 | tzafrir | Where do you see it? |
| 09:03.18 | RadJackson | asterisk MEM% when typing TOP |
| 09:06.08 | RadJackson | this is the primary reason why i would like to separate outgoing from incoming calls on different boxes , at least this will have no impact on incoming calls. |
| 09:06.52 | tzafrir | RadJackson, there are various ways to tracking potential leaks. Do you actually see an increase over time? |
| 09:07.00 | tzafrir | A gradual increase? |
| 09:07.15 | tzafrir | 10% is how much memory? |
| 09:08.24 | RadJackson | gradual increase. over 4 hours of non stop outbound calls it gets > 10% , and the server keeps lagging , even when reloading asterisk , in such case i need to restart :\ |
| 09:08.27 | RadJackson | server config : http://pastebin.com/4AfgQszn |
| 09:10.19 | tzafrir | You see this memory usage with no active calls or with many active calls? |
| 09:11.45 | RadJackson | once rebooting asterisk, MEM is 0% , over 4 hours of non stop outgoing calls it increase gradually to > 10%. After that even when i stop all calls (incoming and outgoing) , the mem doesnt decrease , stuck to its value > 10% |
| 09:14.18 | RadJackson | and the server keeps lagging , peers with 2sec qualify gets LAGGED, sometimes UNREACHABLE , personally i think its the limit of asterisk. |
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| 09:17.12 | RadJackson | tzafrir anyway back to main question , what would you suggest or recommand to run two asterisk boxes (with such load) in the same server? what linux container or vm ? |
| 09:17.44 | tzafrir | There are various other containers that are not docker |
| 09:18.01 | tzafrir | lxc, or the simple systemd ones, or others. |
| 09:18.25 | RadJackson | ok thanks |
| 09:18.26 | tzafrir | You can even simply run two instances of Asterisk from the same installation: |
| 09:18.55 | tzafrir | asterisk -C path/to/asterisk1.conf; asteris -C path/to/asterisk2.conf |
| 09:19.26 | tzafrir | One tricky thing with that is getting all the networking setup (interface binding and such) right |
| 09:19.53 | RadJackson | Okey i will try this. |
| 09:24.57 | Tim_Toady | I'm not sure that just by running 2 separate asterisk processes on the same system will solve much |
| 09:25.10 | Tim_Toady | if you dont find the root of your problem |
| 09:25.43 | RadJackson | Tim_Toady at least i can have a box for incoming calls running < 100 calls. that i wont have to restart , totally independant. |
| 09:27.08 | RadJackson | Tim_Toady i've concluded that the root of the problem is that Asterisk cannot run up to 450 calls (the type of calls we make, ivr/monitoring/recording/agi calls/database manipulation...) |
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| 09:28.34 | SunGod | Hello guys I'm getting "No matching peer for '+4XXXXXXXXXXX' from '178.XX.XX.XX:5060'" on INVITE here is a sip debug and sip.conf http://pastebin.com/dDgNic61 |
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| 09:31.32 | Tim_Toady | RadJackson: or you mmight be hitting the limits of your hardware for the given scenario |
| 09:32.57 | RadJackson | Yes it can be |
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| 11:07.37 | madduck | I understand I should be using _X. as a catchall pattern, but is there a pattern that will match everything including single digits? |
| 11:07.50 | madduck | _. should not be used, right |
| 11:07.53 | madduck | as it matches 0 too |
| 11:08.11 | WIMPy | X! |
| 11:08.35 | WIMPy | But off course it will only match extensions starting with a digit. |
| 11:09.17 | madduck | right, and my problem is that I'd like to simplify the dialplan by matching on s *and* digits |
| 11:09.38 | madduck | so _[a-zA-Z0-9] eh? |
| 11:09.47 | WIMPy | [0-9s]! |
| 11:09.54 | WIMPy | Or that. |
| 11:10.17 | madduck | right. Do I need the !? I am not sure I understood the "immediate matching" yet |
| 11:10.44 | WIMPy | ! matches 0 or more additional character. |
| 11:11.18 | WIMPy | The other difference to . is that it will match as soon as nothing else can match, whereas . will always wait for end of input. |
| 11:11.27 | madduck | right ok |
| 11:11.32 | madduck | thank you I think I get it now |
| 11:11.36 | madduck | let's try and see ;) |
| 11:12.48 | madduck | but _[a-z]! is as bad as _. since it'll match h, t, i and all these specials |
| 11:13.16 | WIMPy | Yes. |
| 11:13.59 | WIMPy | The easiest way to avoid matchin special extensions other than s is to only have a Goto() to another context in that context. |
| 11:14.21 | WIMPy | That wat hite would be searched in that other context and you are safe. |
| 11:14.24 | madduck | right, i was thinking that |
| 11:14.29 | madduck | thank you! |
| 11:14.40 | WIMPy | But you still get the warning. |
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| 13:22.20 | cusco | hey... I know its not the place to ask but I've seen related questions here... |
| 13:22.49 | cusco | so We have this siemens gigaset base, that gets a IP via dhcp, but dfoesn't have web configuration port open |
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| 13:33.23 | RadJackson | slaps [TK]D-Fender |
| 13:35.00 | RadJackson | sorry , wrong slap |
| 13:35.02 | RadJackson | :) |
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| 13:40.45 | c0ldg0ld | |/part |
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| 14:25.59 | cervajs2 | hi, i'm trying webrtc with ast13+chan_sip+sip.js/sipml5+last chrome. with both JS libs i have problem "chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " . sip account has encryption=yes. whole configuration is based on tutorials from wiki.asteirsk.org, sipml5, sip.js. any ideas? |
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| 14:30.36 | qakhan | when Tranfer event shows in queue_log table? |
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| 14:45.53 | newtonr | cervajs2, you have res_srtp loaded? |
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| 14:46.44 | cervajs2 | yew |
| 14:46.46 | cervajs2 | yes |
| 14:47.00 | cervajs2 | libsrtp 1.4.4 |
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| 14:48.22 | cervajs2 | this is strange - [May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=setup:actpass... UNSUPPORTED O R FAILED. |
| 14:49.20 | cervajs2 | i'll paste invite + debug to pastebin |
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| 14:52.04 | cervajs2 | newtonr, http://pastebin.com/VXfpuFGE |
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| 15:35.15 | qakhan | there is Transfer event missing in queue_log, when i transfer the call from 1 agent to other agent, |
| 15:36.13 | qakhan | according to this link http://www.voip-info.org/wiki/view/Asterisk+log+queue_log |
| 15:37.12 | qakhan | Transfer will 100% work when we use asteriks trasfer functionality |
| 15:38.10 | qakhan | i setup blindxfer => # in features.conf |
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| 16:18.55 | madduck | what do people use to block/contain brute-forcers? |
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| 16:20.15 | [TK]D-Fender | fail2ban |
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| 17:09.46 | cervajs2 | madduck, snort.org |
| 17:17.34 | billzeta | Anyone have issue using PJSIP and receiving inbound calls to an endpoint? I have a softphone (zoiper) that can call out but will not receive inbound calls. PJSIP shows the endpoint as unavailable no matter what NAT related settings I apply to the endpoint. res_pjsip flags an error that it can't bind the endpoint to an AOR. The endpoint has a dynamic IP and is behind NAT. Using Chan_Sip i have |
| 17:17.34 | billzeta | no problemsregistering... Both of my SIP trunks are setup via PJSIP and working fine. |
| 17:21.44 | billzeta | ... nevermind ... I just did something and now its binding properly.. wth... |
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| 17:25.31 | madduck | cervajs2: snort is a monitoring suite, last I checked. I just want a policy e.g. 5 invalid extensions within 10 seconds and then the originating IP gets blocked for an hour |
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| 17:49.36 | benlangfeld | Good afternoon everyone. I seem to be experiencing https://issues.asterisk.org/jira/browse/ASTERISK-23425 with Asterisk 11.17.1, with logs and config at https://gist.github.com/benlangfeld/7aa2c7baae607e61b7fc. The issue in brief is that Asterisk doesn't include ICE details in SDP even when apparently configured correctly. Can anyone see what I'm doing wrong? |
| 18:02.07 | newtonr | benlangfeld, I don't see any obvious configuration issue |
| 18:05.51 | benlangfeld | Is there any known issue / heresay that's not in JIRA? |
| 18:05.59 | benlangfeld | I'm having no luck finding anything. |
| 18:08.09 | file | * The libuuid development library is now optional for res_rtp_asterisk. If the |
| 18:08.09 | file | <PROTECTED> |
| 18:08.33 | benlangfeld | ah, that sounds relevant. I'll look into that |
| 18:08.41 | file | pjproject has a dependency on it on some distros, so it was made optional to be less of an impact |
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| 18:53.29 | madduck | how is asterisk trying to write secret.conf to voicemail? |
| 18:53.31 | madduck | [May 19 20:53:00] ERROR[24763][C-00000002]: app_voicemail.c:13617 write_password_to_file: Error writing voicemail password to /srv/asterisk/spool/voicemail/lehel/17/secret.conf |
| 18:53.40 | madduck | but it managed to write the greeting.wav files etc. |
| 18:53.51 | madduck | and I also get this error if I chmod 777 the entire tree! |
| 18:54.09 | madduck | and yes, /srv/asterisk/spool/voicemail/lehel/17 exists |
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| 18:55.06 | epaphus | hi guys OT , looking at the cisco spa303 could anybody tell me if the two ethernet ports that it has if i could use one to connect my pc? |
| 18:56.15 | madduck | wtf, if I touch secret.conf then it works |
| 18:56.29 | madduck | so it fails to create the file but it can write to it? |
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| 19:03.20 | newtonr | madduck, odd , i haven't seen that before |
| 19:03.38 | madduck | yeah neither, been using this feature for years, but now setting up v11 for the first time |
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| 19:04.45 | madduck | i filed https://bugs.debian.org/785743 for now |
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| 19:12.05 | bipolar | Is there a way to disable DTMF detection/retransmission in a call? I have calls coming in on a PRI and going out via FXS ports to some equipment and it needs the original inband DTMF tones from the calling equipment, and to send a few back. |
| 19:14.26 | bipolar | I think I'm missing something simple. |
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| 19:40.18 | newtonr | rmudgett ^ |
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| 19:48.19 | rmudgett | bipolar: Try disabling all DTMF features on that call. No Dial(tTxXkK) flags, DYNAMIC_FEATURES channel variables, chan_dahdi.conf callwaiting or callwaitingcallerid options enabled |
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| 19:49.41 | rmudgett | With all the DTMF features disabled you should get into a chan_dahdi native bridge where the DTMF tones will go straight through. |
| 19:49.48 | benlangfeld | Apologies guys, got pulled away. file, thanks for the suggestion to check libuuid |
| 19:50.09 | benlangfeld | It does appear to have been installed at compile time. |
| 19:50.41 | benlangfeld | Is there any way I can check if ICE support is enabled? Not being a separate module, I'm not sure how I'd check that. |
| 19:50.58 | bipolar | rmudgett: I'll give that a shot. Right now the Dial command has only one option. It just dials an FXS port and sends some dtmf tones for callerID and DNIS info via the D option. I'll see if any of those other variables are getting set. |
| 19:51.03 | bipolar | rmudgett: thanks. |
| 19:51.06 | file | an nm-a res/res_rtp_asterisk.so | grep pjnath |
| 19:51.10 | file | er nm -a |
| 19:51.18 | file | would show it, since those symbols have to exist |
| 19:54.45 | benlangfeld | Not present. |
| 19:55.16 | file | do you also have the development package installed? |
| 19:55.19 | benlangfeld | yes |
| 19:55.31 | benlangfeld | uuid-dev and uuid-runtime |
| 19:55.35 | benlangfeld | This being Ubuntu 14.04 |
| 19:56.01 | file | for some reason then it's likely not being detected and thus not being built in |
| 19:56.34 | benlangfeld | Is missing libuuid the only possible candidate for that? |
| 19:56.41 | benlangfeld | Or could I be missing something else? |
| 19:56.59 | file | that is the only thing that stops it |
| 19:58.25 | benlangfeld | Ok, will recompile and check. |
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| 20:17.15 | benlangfeld | file: Even with libuuid, it's not compiled in: https://gist.github.com/benlangfeld/f7f298fe99bd883ebf8c |
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| 20:29.14 | benlangfeld | I'm attempting in a fresh build environment |
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| 20:37.13 | benlangfeld | file: still no luck :( https://gist.github.com/benlangfeld/f7f298fe99bd883ebf8c |
| 20:37.24 | benlangfeld | Am I still doing something wrong or should I file a JIRA? |
| 20:42.36 | benlangfeld | Note that this is Asterisk 11.17.1, not 13, so no chan_pjsip. |
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| 21:47.50 | bipolar | rmudgett: If I disable all the feature codes in the system, globally, it works. Is there any way to do this inside of a context? |
| 21:48.25 | bipolar | This system is also going to handle normal voice traphic and phones. |
| 21:49.53 | rmudgett | bipolar: The Dial options and DYNAMIC_FEATURE options can be done per call. The chan_dahdi.conf options cannot. |
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| 21:55.13 | bipolar | rmudgett: Scratch that. It was working when I was using a single line like: exten => 5555551234,1,Dial(DAHDI/97,3,D(*${CALLERID(ANI-num)}*51234*)) |
| 21:56.02 | bipolar | but as soon as I pass it to my other context that does some other things with it (call recording, etc.) it starts reading the dtmf tones again. :( |
| 21:56.27 | rmudgett | Call recording precludes native bridging. |
| 21:56.41 | bipolar | Damnit. |
| 21:56.48 | rmudgett | Asterisk has an interest in the media then. |
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| 21:57.34 | bipolar | I just wish I could keep it from trying to 'fix' the dtmf tones. |
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| 22:02.51 | bipolar | rmudgett: this stuff: http://pastebin.com/Y0R38z8U |
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| 22:04.17 | rmudgett | The DTMF digits would need to be longer than 80ms to avoid the emulation that stretches the digits to 80ms |
| 22:05.11 | rmudgett | That behavior is part of the bridging core and cannot be turned off. Native bridging bypasses that path. |
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| 22:06.51 | bipolar | rmudgett: not possible. It's not a phone sending them, it's an alarm panel. Also, there is the option mindtmfduration in asterisk.conf. I have it set to 20. The first '9' it sees is only 51ms and it passes though. Not sure why it's not letting the others though. |
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| 23:05.10 | bipolar | rmudgett: Looks like if I disable the *in-call* feature codes, and don't try to record the channel, it uses a native bridge and doesn't mess with the inband dtmf tones. Thanks for your help! |
| 23:06.16 | bipolar | I do wish there was a way to get it to pass though dtmf tones without molesting them on a per-exten basis, but at least it's functional this way. |
| 23:07.21 | WIMPy | monitoring and features are dialplan thing, so that's per extension, isn't it? |
| 23:07.23 | rmudgett | That would be your dialplan for per exten |
| 23:07.46 | rmudgett | WIMPy: yes |
| 23:08.21 | WIMPy | But that reminds me that I need to find out why DTMF detection can disable audio... |
| 23:10.43 | bipolar | WIMPy: it seems that it covers up the audio to hide the DTMF tone it's pulling out of band. At least that's what I gather. I could be wrong. |
| 23:12.31 | [TK]D-Fender | bipolar, possible option : Macro/gosub. originate another channel that will bridge/chanspy with the target leg and pass it on that way. |
| 23:12.40 | bipolar | I really could use a SET(DONTMOLESTDTMF=1) to just turn the whole thing off in a context. |
| 23:12.48 | [TK]D-Fender | bipolar, I don't believe it will break the native bridge in that case |
| 23:13.04 | bipolar | [TK]D-Fender: hmmm... |
| 23:13.29 | bipolar | [TK]D-Fender: It would still be a native bridge and unable to do things like recording though, right? |
| 23:14.00 | [TK]D-Fender | I'm wondering if you could use chanspy as a means of doing the recording... |
| 23:14.36 | WIMPy | Might work. |
| 23:21.38 | rmudgett | WIMPy: That won't work. chanspy makes Asterisk have an interest in the media so native bridging won't happen |
| 23:22.02 | WIMPy | Yes. Makes sense. |
| 23:23.12 | WIMPy | The funny thing is that my issue is the exact opposite. It's DTMF on a native bridge. |
| 23:23.32 | [TK]D-Fender | rmudgett, the native bridge has already HAPPENED however |
| 23:23.42 | bipolar | WIMPy: lol. just record it and it won't be native any more. :P |
| 23:24.09 | WIMPy | Yes, but I want native to have the extra features. |
| 23:25.19 | rmudgett | Attaching chan_spy to a channel in native bridging will break the native bridge to a normal bridge. |
| 23:25.33 | [TK]D-Fender | if you say so... |
| 23:25.39 | rmudgett | convert the native bridge to a normal bridge |
| 23:26.02 | WIMPy | So ATM I can have either this or that. And I'm not sure for how long it has been broken. But I'm pretty sure I used to have both. |
| 23:26.38 | rmudgett | Its more obvious what happens in v13 to the bridging modes |
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| 23:39.41 | bipolar | Is there a way to disable the in-call feature codes without doing it globlaly? That would make this more robust. |
| 23:40.00 | WIMPy | Don't enable them. |
| 23:40.48 | bipolar | I mean, if the feature codes are enabled globally, is there a way to disable them for this case? |
| 23:42.37 | rmudgett | What specific feature codes are you enabling globally? |
| 23:42.46 | rmudgett | And how |
| 23:44.59 | rmudgett | Feature codes are enabled as Dial options. This is per call. DYNAMIC_FEATURES is a channel variable that you can unset. |
| 23:50.43 | bipolar | rmudgett: for instance.... in features.conf - blindxfer=## |
| 23:51.42 | rmudgett | That just says what keys trigger blind transfer. The Dial(tT) enables it on the call. |
| 23:51.43 | WIMPy | You still need to enable that in Dial() with t/T. |
| 23:52.05 | bipolar | yeah... wtf |
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| 23:52.10 | bipolar | hold on.... |
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| 23:58.06 | bipolar | It's a custom feature that's breaking native bridging. Makes sense. |
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