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| 00:59.24 | igcewieling | I'll wait until people like muks find most TLS/SRTP problems before I try it again. |
| 01:00.21 | igcewieling | thanks muks for his sacrifice. |
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| 07:10.29 | muks | igcewieling: asterisk TLS (pjsip) and SRTP with 128 bit key works well for me with multiple clients. the TLS certificate validation problem is a client-side problem that is unrelated to asterisk. the SRTP 256 bit key issue may be an asterisk or client problem. |
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| 09:59.30 | gypsymauro | hi |
| 10:02.09 | gypsymauro | suppose I've separated my netowrks with VLAN, I saw that there are phones with more than a NIC so I can attach PC after the SIP terminal , in the cases of few connection ports, it's possibile to have VLAN on phones and separate the traffic there? on the switch port both VLANs and using the corrispondent on each device? (terminal and PC) |
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| 13:50.59 | ilius | gypsymauro, the answer is yes. Polycom and many other phones can join a specific voip vlan on their uplink port with LLDP communication from the switch, then, the PC port is just VLAN 1 or whatever you want it to be. Of course, it requires a pretty smart switch. |
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| 13:55.52 | ilius | Oh, I guess they left. Meh. |
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| 15:00.30 | ilius | I believe I've tracked my order-of-magnitude increase in cpu usage increase for rtp starting with this change: https://gerrit.asterisk.org/c/asterisk/+/5341 |
| 15:01.04 | ilius | It basically wraps practically every operation in an ao2 lock. |
| 15:02.21 | igcewieling | ilius: you tried transcode_via_sln=no in asterisk.conf ? |
| 15:03.04 | ilius | Adding more cpu cores only increases the load on the machine because it locks across all cores. It basically made it into a single threaded multi-threaded application. |
| 15:04.52 | ilius | This instance does not transcode. Everything is mulaw. But I can try that just to knock that possibility out. |
| 15:05.53 | ilius | I looked at Kamailio, but it doesn't do rtp. I'd have to use rtpengine or some other thing. |
| 15:06.13 | igcewieling | ilius: *nod* It is a simple enough change to try. |
| 15:10.54 | ilius | I thought it was pjsip, and it isn't. Not directly, at least. The same thing happens on chan_sip after Asterisk 13.23. According to Mudgett's notes, the introduction of all of the thread locking was because pjsip required it, and it wasn't necessary before. |
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| 15:12.56 | cid254 | Hi, i m trying to use cisco 7961 with freepbx 15 (asterisk 16) but they don't register. I've tried both legacy chan sip and pjsip, also tried to switch from udp transportlayer to tcp without luck. any hint? |
| 15:13.15 | cid254 | also i set nat to never |
| 15:13.16 | ilius | Okay, with transcodev_via_sln = no, I am still seeing the same CPU usage. 1 call is consuming about 15-20% of a virtualized Haswell core at 3.8Ghz. |
| 15:16.11 | ilius | Oh, somthing else was going on. It's really about 5-15% of the core. At this rate, this machine can relay rtp for maybe 7-20 calls before load goes > 1. |
| 15:16.57 | igcewieling | ilius: thanks for letting me know. I process 50+ calls at a time with a system load under 1 |
| 15:18.04 | ilius | I'm going to send some production calls through just to be sure. |
| 15:18.31 | ilius | thanks igcewieling. I wonder if maybe qemu+kvm isn't friendly to constant thread locks. I dunno. |
| 15:19.21 | igcewieling | no idea, but it might be good to mention at the start |
| 15:19.37 | file | what's the output of "core show settings"? |
| 15:19.46 | seanbright | file: couldn't take it anymore, eh? |
| 15:19.51 | seanbright | i've been waiting |
| 15:20.11 | file | seanbright: are you thinking what I'm thinking? |
| 15:20.22 | seanbright | file: muffins? absolutely |
| 15:20.29 | file | seanbright: root beer! |
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| 15:25.09 | ilius | https://pastebin.com/raw/ggHvV6w7 <= core show settings (keep in mind that this is my downgrade to 13 > 13.23 to test if chan_sip experiences the same load.) |
| 15:25.33 | file | don't enable DEBUG_THREADS. |
| 15:25.45 | seanbright | game. set. match. |
| 15:25.47 | seanbright | heh |
| 15:25.57 | ilius | I have that enabled to see what's going on :) |
| 15:26.13 | file | then you've turned Asterisk into, as you said, a single threaded multi threaded application |
| 15:27.50 | ilius | Wow. Okay. I'll try turning that off again. How do I debug which threads are pegging then? |
| 15:28.03 | file | a profiler most likely |
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| 15:38.16 | ilius | https://pastebin.com/raw/AqU3kiqb <= this is the core show settings of my current primary production asterisk dmz instance that sees the same rtp cpu load. |
| 15:41.36 | ilius | I'll turn off Transcode via SLIN on that instance. Other than that, I dunno... |
| 15:42.18 | ilius | It currently has a load of 1.1 with about 50% with 13 active calls. |
| 15:43.11 | ilius | s/50%/50% cpu/ |
| 16:12.11 | avb | igcewieling: Samot: finally solved my 'weird issue' of the other day |
| 16:12.13 | ilius | Does Asterisk still do the sln translation internally, even when the channel looks like this? https://pastebin.com/raw/5wkpXi8F |
| 16:12.45 | seanbright | mixmonitor needs signed linear |
| 16:12.53 | seanbright | so, short answer: yes |
| 16:13.12 | seanbright | if you are using mixmonitor you will always transcode to signed linear |
| 16:13.14 | avb | most likely ops team have silently upgraded vmware on where asterisk been running and that caused network drop the packets somehow |
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| 16:13.56 | avb | ops have installed new vmware server, restored the backups and magically everything went back to normal operation |
| 16:14.40 | ilius | seanbright, wow, thank you. does mixmonitor still do that if transcode_via_sln=no? |
| 16:14.56 | seanbright | i don't know what that option is supposed to do |
| 16:15.31 | seanbright | ulaw <-> sln is fairly cheap though, so i can't imagine that transcoding is your problem |
| 16:16.07 | ilius | how do I just dump the ulaw streams to a file without transcoding? Do I have to use the Monitor() app and mix it myself? |
| 16:16.50 | seanbright | yes |
| 16:16.52 | igcewieling | ilius: I have never been sure transcode_via_sln caused spikes in CPU usage on my servers, so don't assume that is truly the case. |
| 16:16.54 | ilius | Ok. I'm waiting for this last production call to finish and I"m going to set transcode_via_sln=no on that instance, hopefully watch the usage go down. |
| 16:18.04 | seanbright | i don't see how that option would affect you at all |
| 16:18.13 | seanbright | you're transcoding TO signed linear |
| 16:18.29 | seanbright | if you were transcoding to something else (ulaw -> g729 for example) then it might |
| 16:19.12 | ilius | yeah, i've tried to remove all transcoding everywhere just to remove that variable from the equation |
| 16:19.51 | ilius | i guess mixmonitor probably reencodes sln back to ulaw |
| 16:19.58 | seanbright | correct |
| 16:20.12 | seanbright | ulaw -> slin, mix as slin, slin -> ulaw |
| 16:22.30 | ilius | ok. after this call that has been going on for an hour finishes, and I switch to transcode_via_sln=no, i'm going to explore using Monitor() instead... if it indeed doesn't do anything with the stream. |
| 16:22.48 | igcewieling | file: thought you might like this: https://pastebin.com/BuYnmCQ5 |
| 16:23.23 | igcewieling | You are not alone! |
| 16:23.34 | *** join/#asterisk Philomena420 (~ryzen@d4-50-149-85.nap.wideopenwest.com) |
| 16:23.51 | file | ha |
| 16:25.42 | Philomena420 | anyone willing to answer a question about access to emergency services? |
| 16:28.25 | Samot | What's the question? |
| 16:29.15 | Philomena420 | so my voip provider doesnt have 911. Is there some reading somewhere on providers or is there a service out there that provides it for free? |
| 16:30.13 | Philomena420 | we have cell phones of course but this was something i was considering having as a backup for my office phone at home if somethere were to happen or my kids were to godforbid get hurt |
| 16:33.00 | Samot | Are you in the US? |
| 16:33.04 | Philomena420 | yup |
| 16:33.20 | Samot | Well by Jan 2021 they have no choice but to give you 911 |
| 16:33.44 | Philomena420 | so all providers have to have it even if you dont have a DID? |
| 16:34.02 | Samot | Yup. |
| 16:34.15 | Samot | Is this new or old service? |
| 16:34.41 | Philomena420 | i havent had this provider long |
| 16:34.51 | Samot | New as in since Feb 16th. |
| 16:34.56 | Philomena420 | yes |
| 16:35.19 | Samot | Well I'd find a new provider. |
| 16:35.31 | Samot | Or ask them when they are going to be in compliance with the new laws. |
| 16:35.54 | Philomena420 | so at the moment there arent any free providers of the service or will there be i guess is what im asking? |
| 16:36.16 | Samot | At the moment there are now new laws that state 911 is a must deliver service. |
| 16:36.23 | Philomena420 | or at least the ability to supplement the plan with 911 services at no upcharge |
| 16:36.35 | Samot | They have until Jan 2021 to comply with their current users. |
| 16:36.58 | Philomena420 | got it...thank you...one last question |
| 16:37.12 | Samot | You need to ask your provider when they are complying with the new laws. If they say "What new laws" get another provider. |
| 16:37.25 | Philomena420 | how does 911 get routed from my server then? sorry for the newbie line of questioning |
| 16:37.35 | Samot | What server? |
| 16:38.00 | Philomena420 | my asterisk server if i were to set it up with 911 via my provider or is it handled by my provider when i give them my address |
| 16:38.39 | Samot | OK so you have an ITSP that you are using for outbound. |
| 16:38.55 | Philomena420 | yes |
| 16:39.41 | Samot | OK so you're not getting it for free. |
| 16:40.14 | Samot | Any ITSP offering 911 calling is going to have a cost. |
| 16:40.20 | Philomena420 | i currently pay a certain amount per call if thats what you mean |
| 16:40.22 | Philomena420 | ahhh ic |
| 16:40.25 | Samot | You need a DID and to register your address/location. |
| 16:40.57 | Samot | So let me enlighten you on something. |
| 16:41.02 | Philomena420 | please |
| 16:41.04 | Samot | Because this is very important. |
| 16:41.12 | *** topic/#asterisk by kharwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.34.0 (2020/06/11) 16.11.1 (2020/06/16) Standard: 17.5.1 (2020/06/16); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 16:41.21 | Samot | You've installed a PBX system in your home. You are the PBX installer/admin. |
| 16:41.40 | Samot | Understand, these new laws actually make YOU 100% responsible to make YOUR PBX comply to them. |
| 16:41.55 | Samot | So actually the ITSP is absolved. |
| 16:42.21 | Samot | Because the new laws related to the actual MLTS/PBX system in question the devices connect to. |
| 16:42.42 | Philomena420 | so since i went diy instead of having someone else act as the "admin" |
| 16:42.49 | Philomena420 | im responsible |
| 16:43.17 | Samot | Yes, you could have someone else be the admin/installer but no one is doing that for free and if they are, they don't understand the new laws. |
| 16:43.30 | igcewieling | Getting a DID with 911 service is easy, there is no excuse not to. |
| 16:43.46 | Samot | So your PBX must be able to dial 911, allow calls backs to the DID presented to 911 and have a registered address with them. |
| 16:43.55 | Samot | Which means you need to buy a DID |
| 16:44.10 | Samot | You need to get 911 services from your provider so they can register your address. |
| 16:44.27 | Samot | You need to make sure your PBX can dial 911 with no prefixes or special routing. |
| 16:44.37 | Samot | It must dial to the PSAP |
| 16:44.49 | Philomena420 | should i do this through my current provider i is it wise to shop around for a DID/another provider |
| 16:44.55 | Philomena420 | or* |
| 16:45.02 | Samot | You have a PBX |
| 16:45.12 | Philomena420 | i mean its a rapsberry pi |
| 16:45.22 | Samot | It doesn't matter what the server is |
| 16:45.27 | Philomena420 | ha yeah |
| 16:45.30 | Samot | It is a Multi-Line Telephone System. |
| 16:45.46 | Samot | The new federal laws apply specifically to that type of system/service. |
| 16:45.55 | Samot | And the installer/admins of them. |
| 16:47.07 | Samot | See many DIYers/VoIPers are going to soon realize their home PBX systems they've had for ever will be out of compliance of the federal laws. |
| 16:47.32 | Samot | And that realization may come in the form of a fine or something. |
| 16:47.37 | Philomena420 | so right now im safe because im not paying for the 911 service, i only dial out with my providers numbers |
| 16:47.44 | Samot | No. |
| 16:47.50 | Samot | YOU INSTALLED a PBX SYSTEM |
| 16:48.01 | Samot | That is all that matters to the laws. |
| 16:48.06 | Samot | They don't care about who your provider is. |
| 16:48.12 | Samot | They care that the PBX is in compliance. |
| 16:48.33 | Philomena420 | could you provide me the reading or is this accessable somewhere so i understand the laws |
| 16:49.12 | Samot | The FCC website. |
| 16:49.28 | Samot | Google: Kari's Law and Ray Baum's Act Section 503 |
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| 16:54.59 | Philomena420 | reading now...thanks |
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| 17:08.32 | Philomena420 | so for the time being is this ok? www.ipcomms.net/911labels |
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| 17:45.06 | zelele | Hi everybody. |
| 17:47.05 | zelele | Does someone had any problem with Asterisk 16 to put CDR_ODBC to work ? Because I already did It to work with until Asterisk 13, but now with this 16 version I'm receiving cdr failed: |
| 17:47.48 | zelele | In the message It gives me error to insert record on table |
| 17:47.49 | zelele | INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence) VALUES ({ts '2020-06-16 01:53:47'},?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?) |
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| 18:27.30 | zelele | I am asking you about this possibility, as I have usually installed databases on MySQL instances normally up to version 13. |
| 18:27.30 | zelele | But when I went to use the same scheme in version 16, right away I came across this possible error. |
| 18:55.47 | igcewieling | Did you read the UPGRADE.txt files? |
| 18:56.02 | igcewieling | Did you check the schema which comes with Asterisk 16? |
| 19:02.20 | zelele | I did not read the entire UPGRADE.txt document as I am using a clean installation of Asterisk 16.9.0. But as far as I could verify, there were no significant changes from version 13 to version 16 in this regard. Unless I'm wrong. |
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| 19:03.22 | zelele | To build the table in the "cdr" database, I used the schema that came in the file contrib/realtime/mysql/mysql_cdr.sql |
| 19:03.28 | zelele | Am I right ? |
| 19:05.12 | zelele | The difference that I could observe between the scheme I used previously and this scheme that I commented on, are the positions of two or three columns. Even so, I followed the schematic of the table located in the contrib directory. |
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| 19:24.16 | ilius | The only thing I ran into migrating from cdr_mysql to cdr_odbc, sip to pjsip, and from 13 to 16 is that the `answer` and `end` are no longer populated, which I remedied with a default calculation from billsec and duration. |
| 19:25.07 | ilius | If you can get the actual error from mysql, you can probably pinpoint what the problem is... does odbc abstract that error away? |
| 20:15.50 | hardwire | Using hangup_handler_push right now with some static dialplan. Moving this app over to Stasis/ARI and I'm not quite sure how to get the tech hangup_cause event if I originate via ARI PUSH channels. Anybody tackled this? |
| 20:16.10 | hardwire | Trying to avoid dialing a local channel in order to do this. |
| 20:21.27 | igcewieling | not exactly sure what you are doing, but have you looked at generating a manager event inside your hanguphandler, GenCELUserEvent or whatever it is called? |
| 20:23.49 | hardwire | Yeh that may be how to deal with this. I was trying to avoid having any extra dialplan. The app originates outbound and other than handling hangup causes it doesn't need anything else. I will try setting hangup_handler_push in the origination variables to see if it attempts the handler. |
| 20:25.53 | hardwire | I mainly just need the SIP codes from the upstream when rejecting a call. |
| 20:29.19 | hardwire | Hopefully I can just feed it back to Stasis that way;) |
| 20:32.38 | hardwire | yeh.. completely ignores it. Going to have to use Local as tech. |
| 20:33.01 | hardwire | That'll double my channel count. |
| 20:33.06 | igcewieling | This won't help you, but I do a Set(MASTER_CHANNEL(outchan)=${CHANNEL(name)})) in a pre-dial handler so I can use HANGUPCAUSE on the dialing channel to get the outbound leg hangupcause. |
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| 20:33.39 | hardwire | yeh. I can get the asterisk hangup cause per leg just fine ;_ |
| 20:33.44 | hardwire | Just need that tech cause. |
| 20:34.48 | igcewieling | I meant something like ${HANGUPCAUSE(${outchan,tech})} which should allow you to extract the sip code if using a hanguphandler with a predial handler |
| 20:34.51 | hardwire | I also feel like I'm doing something silly. I'm originating via REST and not passing the call to any application or anything. Just handling the call as is. Perhaps this is perilous.. but it also just seems clean. |
| 20:35.55 | igcewieling | The only "clean" dialplan is one which doesn't do anything useful. Dialplans are ugly, no way around that. |
| 20:36.20 | hardwire | igcewieling: Right. I just won't be originating PJSIP via ARI. I'll have to head through Local first. |
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| 20:48.47 | hardwire | igcewieling: this should work fine. just using DIALEDPEERNAME does the trick |
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| 22:12.10 | ilius | Hrm, res_monitor is an "extended" module in 16.8-cert2. I never noticed it was deprecated. |
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| 22:13.14 | ilius | Whew: https://www.mail-archive.com/asterisk-dev@lists.digium.com/msg44886.html |
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| 22:22.01 | hardwire | It appears as though UserEvent's emitted in hangup handler dialplan set up by way of b(...) dialplan options don't make their way back to the Stasis app. |
| 22:22.13 | hardwire | Setting the hangup handler pre-dial however does work. |
| 22:22.48 | hardwire | the UserEvent happens - just doesn't hit the right app bus |
| 22:23.25 | igcewieling | hardwire: sounds like possible progress |
| 22:24.56 | hardwire | I'm thinking just use HEP at this point. |
| 22:25.01 | hardwire | TBH |
| 22:28.17 | hardwire | <PROTECTED> |
| 22:28.17 | hardwire | <PROTECTED> |
| 22:28.17 | hardwire | <PROTECTED> |
| 22:28.17 | hardwire | <PROTECTED> |
| 22:28.21 | hardwire | bout as good as it gets. |
| 22:29.07 | hardwire | I found UserEvent to be a bit cryptic as well. UserEvent(name,thing:1,thing:2,thing:3). Docs need to reflect this abit better. |
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