| 00:00.25 | jwpierce3 | [TK]D-Fender, I got my queue to restart by removing n from extensions.conf Queue(), and by setting timeoutrestart=yes in queues.conf. But from what I read the queue timeout is set by the Queue() application in extensions.conf. Queue(name,options,,,timeout) |
| 00:00.49 | [TK]D-Fender | you have failed to read the sample config |
| 00:00.53 | [TK]D-Fender | Do so again |
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| 00:31.05 | jwpierce3 | [TK]D-Fender, thank you. timeout=0 timeoutpriority=app |
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| 08:15.33 | Lope | when I change my SIP bindport, will that port get sent to the registrar when I send then register requests with register => ? |
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| 08:31.22 | Samot | No. |
| 08:31.30 | Samot | Well yes. |
| 08:31.38 | Samot | It's your listening porting. |
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| 08:38.59 | Rasputin3711 | Does anybody know how to calculate load on a server? |
| 08:40.20 | Samot | w |
| 08:40.50 | Samot | What are you trying to determine? |
| 08:42.51 | Rasputin3711 | For example: I need 50 users, max 20 concurent calls, 2 queues, 1 queue (max 5 users) with call recrod. What cpu&ram do i need for the system? |
| 08:45.24 | Samot | 2GB of RAM should do that. |
| 08:46.11 | Rasputin3711 | 1 core/2 core? |
| 08:46.37 | Samot | 2 core... |
| 08:47.29 | Samot | Is this your first time doing this? |
| 08:48.22 | Rasputin3711 | Yes,it's |
| 08:50.23 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 08:50.25 | Haris | hello all |
| 08:52.17 | Samot | Rasputin3711: If this is your first time, you've read up on thingslike Queues and call recording, correct? |
| 08:56.55 | *** join/#asterisk aness (~aness@cm-84.215.174.224.getinternet.no) |
| 08:59.41 | Lope | ERROR[70]: tcptls.c:912 ast_tcptls_client_start: Unable to connect SIP socket to x.x.x.x:54283: Connection timed out |
| 09:00.02 | Lope | Asterisk is trying to connect to my softphone that WAS connected yesterday, before I shut down the asterisk VM. |
| 09:00.25 | Lope | That softphone is no longer connected etc, I've since restarted asterisk. How can I flush whatever cache it's looking at? |
| 09:01.03 | Haris | I'v installed 11.7 on my local LAN machine. I have no ssl certificate. can I configure it such that sip calls would work locally ? |
| 09:01.04 | Samot | Are you meaning to use both TCP and TLS? |
| 09:01.06 | Haris | on LAN |
| 09:01.22 | Samot | Haris: Yes. |
| 09:01.31 | Haris | I have configured 2 extentions in sip.conf. sip user registration fails owing to ssl certificate at this point |
| 09:01.42 | Samot | Haris: Extensions can call locally within the PBX but I would recommend upgrading your Asterisk. |
| 09:02.05 | Samot | Haris: You need to make sure you don't have settings configured to use TLS. |
| 09:02.37 | Haris | I'm on ubuntu 14.4 LTS. this is the pkg version I saw from apt-get/aptitude |
| 09:02.43 | Lope | how can I flush my SIP peer cache? |
| 09:02.53 | Haris | hmm |
| 09:03.11 | Haris | if I provide asterisk witha self generated ssl certificate, would it accept it ? |
| 09:03.55 | Samot | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 09:04.07 | Samot | https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 09:04.30 | Samot | Lope: Did you restart/reload Asterisk? |
| 09:04.56 | Lope | I tried `sip prune realtime peer all`, but that's not finding anything to prune. when I run `sip show peers` I see one of my sip peers is showing an IP address and port that it had yesterday. How can I clear that? |
| 09:05.12 | Lope | Yes, I've restarted asterisk, also done `sip reload`. |
| 09:05.18 | Samot | sip unregister <peer> |
| 09:05.29 | Samot | Asterisk doesn't hold registrations after you've turned it off. |
| 09:05.35 | Lope | oh cool, yeah JUST found that right now. |
| 09:06.01 | Lope | err, are you sure. I recall seeing an option for a DB to store the sip peers between restarts. |
| 09:06.26 | *** join/#asterisk zamba (marius@flage.org) |
| 09:06.52 | Samot | Asterisk doesn't initiate the registration to the endpoint. |
| 09:07.41 | Lope | well, I'm using TCP transport. and the module that was complaining was tcptls: ERROR[70]: tcptls.c:912 ast_tcptls_client_start: Unable to connect SIP socket to x.x.x.x:54283: Connection timed out |
| 09:07.42 | Samot | Also |
| 09:07.49 | Lope | Have you had experience with that? |
| 09:07.53 | Samot | Are you using TLS? |
| 09:07.58 | Lope | not that I know of hehe. |
| 09:08.02 | Lope | But I am using TCP transport. |
| 09:08.09 | Samot | OK. So that error is very clear. |
| 09:08.16 | Samot | It's trying to use TLS over TCP. |
| 09:08.35 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 09:08.37 | Samot | Also, if it's loading from the database... |
| 09:08.41 | Lope | that's weird because my SIP peer definition is dynamic, type friend etc. |
| 09:08.50 | Samot | It's expecting that x.x.x.x to connect..... |
| 09:09.07 | Samot | Remove the entry from the sipregs database base. |
| 09:09.13 | Samot | Run a SIP debug. |
| 09:09.17 | Samot | Have you done that? |
| 09:09.17 | Lope | yeah. I've not experienced this on 13.6. I've downgraded to 13.1. |
| 09:09.32 | Samot | Why would you downgrade? |
| 09:09.35 | Lope | `sip unregister` worked. |
| 09:09.47 | Samot | Especially so far back?! |
| 09:09.57 | Lope | Well Since ubuntu 16.04 has asterisk 13.1 I figured I don't have to compile manually now, and I can get security updates etc. |
| 09:10.02 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 09:10.07 | Lope | Is 13.1 very old? |
| 09:10.08 | Lope | hehe |
| 09:10.09 | Samot | Dude. |
| 09:10.26 | Samot | 13.6 WAS part of the security updates to Asterisk. |
| 09:10.32 | Samot | It's on 13.9.1 |
| 09:10.39 | Lope | oh lol, so why is ubuntu stagnated on 13.6? |
| 09:10.44 | Lope | I mean ubuntu stagnated on 13.1 |
| 09:10.45 | Samot | Just like 13.9.1 was a fix to an issue with 13.9.0 |
| 09:10.57 | Samot | Because they are packages. |
| 09:11.05 | Samot | Someone has to maintain them. |
| 09:11.13 | Lope | So you reckon compiling is unavoidable? |
| 09:11.22 | Samot | And distro packages like that are ALWAYS behind. |
| 09:11.37 | Lope | yeah. It seems most of the hardcores around here compile. |
| 09:12.00 | Lope | So I wonder if this is a bug. |
| 09:13.01 | Lope | Probably is. It shouldn't be doing this, like a day after the phone is gone, after a restart etc. The asterisk service was stopped etc. |
| 09:13.27 | Lope | Okay I'll compile. |
| 09:13.42 | Lope | I'll just ignore this error if it comes up again until I've got time to compile. |
| 09:14.05 | Lope | Was TCP transport a relatively new feature around the time of 13.1? |
| 09:14.25 | Lope | maybe I can just disable TLS somewhere... |
| 09:14.44 | Samot | No. |
| 09:14.55 | Lope | the funny thing is on other distros you get asterisk 10 and 11 hehe |
| 09:15.03 | Samot | If you aren't using TLS then yes, you should disable references to use it. |
| 09:15.27 | Lope | hmm. where would the references be? I don't have any TLS stuff in my sip.conf |
| 09:18.04 | Lope | I found this: /http.conf:71:;tlsenable=yes |
| 09:18.18 | Lope | I don't know what HTTP.conf is all about? |
| 09:18.33 | Lope | ./xmpp.conf:23:;usetls=yes |
| 09:18.41 | Lope | ./udptl.conf:8:udptlstart=4000 |
| 09:18.52 | Lope | that's all. |
| 09:19.31 | Lope | Looks all unrelated to TCPTLS |
| 09:20.11 | Lope | Nevermind I'll just proceed as above. Will ignore then move to a newer version again. |
| 09:32.16 | *** join/#asterisk wil_syd (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 09:41.44 | Lope | BTW The only thing I have that might be VAGUELY interpreted to be related to TCPTLS is in my sip.conf : tcpenable=yes |
| 09:41.45 | Lope | hehe |
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| 09:59.03 | Lope | ------ |
| 09:59.27 | Lope | This is the first time I've ever tried to dial a phone number, using a VoIP provider, from * |
| 09:59.38 | Lope | Previously I've only dialed softphones from * |
| 10:00.24 | Lope | ohh, I forgot the SIP part. |
| 10:03.14 | Lope | I'm registering with my VoIP provider, and I can receive calls from them. (So I've got a register=>... and a context in sip.conf and extensions.conf for receiving the dall) that's working. |
| 10:03.25 | Lope | Now I want to Dial using the sip provider. |
| 10:04.02 | Lope | So I've got a context where I receive a call from my softphone, then I tried Dial(SIP/555@my.voipprovider.com,30) |
| 10:04.50 | Lope | chan_sip.c:23212 handle_response_invite: Failed to authenticate on INVITE to '<sip:mysoftphone@myasteriskIP>;tag=f9k34f94k34g' |
| 10:06.49 | Lope | I have host=my.voiprovider.com in sip.conf |
| 10:07.36 | Lope | The softphone can make and receive calls. So it must be some kind of issue with authenticating dialing with my VoIP provider. |
| 10:10.31 | *** join/#asterisk obelixBE (~obelix@178-117-79-17.access.telenet.be) |
| 10:14.57 | Lope | when I do a SIP debug it says read from sip provider ip, SIP/2.0 401 Unauthorized |
| 10:15.30 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:16.10 | Lope | `sip show registry` shows that * is registered to the voip provider, with the username correctly. |
| 10:16.20 | Lope | State: registered |
| 10:17.08 | Lope | will try add a username= |
| 10:17.17 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:2d81:a5b1:22da:79d) |
| 10:17.56 | Lope | err, defaultuser= |
| 10:18.35 | Lope | still failed. |
| 10:25.13 | Lope | Here's my sip.conf section for the voip provider |
| 10:25.14 | Lope | http://codepad.org/henapqd5 |
| 10:30.04 | jwpierce3 | when compiling in pjproject, asterisk isn't going to use any sound or codecs right? |
| 10:30.10 | Lope | Maybe the problem is my call to the dial app. because it says Failed to authenticate on INVITE to sip:mysoftphone@asteriskIP. That sounds like my softphone account is trying to authenticate to dial at my voip provider. which doesn't sound right. |
| 10:30.57 | Lope | jwpierce3: I don't know, why does it matter? just follow the instructions and compile it? |
| 10:33.13 | jwpierce3 | Lope, just trying to determine exactly what is needed. Why would be nice, :). I'm adjusting a gentoo ebuild to acommodate for pjsip |
| 10:36.45 | Lope | oh, why not follow the instructions in a VM/container of debian or something, and see what packages get pulled? |
| 10:37.10 | Lope | If you speak to jkroon: I think he uses Gentoo. |
| 10:37.58 | jkroon | i do. |
| 10:38.30 | jkroon | jwpierce3, please join #gentoo-voip |
| 10:38.51 | Lope | jkroon: do you think you could please help me figure out my invite/authentication is failing when I try dial? I'm compiling some debug information and will post it in 2 mins? |
| 10:38.54 | jkroon | jwpierce3, please join #gentoo-voip - i help the maintainer with the package, but we normally discuss those in #gentoo-voip. |
| 10:39.03 | jkroon | not at the moment no sorry Lope |
| 10:39.10 | Lope | okay, all good. |
| 10:41.27 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 10:43.01 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:04.02 | jwpierce3 | Lope, post it, I'll look at it |
| 11:09.29 | Haris | does this basic setup of asterisk allow video call when two basic sip extensions are added ? |
| 11:12.35 | Lope | http://codepad.org/Pl9NsF6b |
| 11:12.39 | Lope | thanks a lot jwpierce3 |
| 11:13.04 | Rasputin3711 | remove password ) |
| 11:13.11 | Lope | so guys ^^ that's my config. I can receive calls from the voip provider but I can't dial |
| 11:14.09 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 11:14.25 | Lope | lol, after all that effort hahaha |
| 11:14.35 | Lope | I guess I have to change my password hahaha |
| 11:14.49 | Lope | the perils of copy and paste. |
| 11:15.22 | jwpierce3 | you changed the url and forgot the username and password??? :P |
| 11:15.35 | Lope | shit happens :P |
| 11:15.47 | jwpierce3 | to the best of us |
| 11:16.07 | Lope | obfuscated the fuck out of everything else lol. |
| 11:17.35 | Lope | Any ideas why I can't dial with the softphone account? |
| 11:19.01 | jwpierce3 | I'm looking |
| 11:19.45 | jwpierce3 | who's your provider? |
| 11:20.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:21.24 | Lope | it's sip.nexus.co.za |
| 11:21.38 | jwpierce3 | type=friend |
| 11:22.00 | jwpierce3 | do they have an option for ip base authenication? |
| 11:22.11 | Lope | oh, I'll try that. |
| 11:22.18 | Lope | I don't know. I made up the config. |
| 11:22.35 | jwpierce3 | log into your nexus account and add your server |
| 11:22.39 | jwpierce3 | server's ip |
| 11:22.52 | Lope | their control panel sucks, it doens't offer that ability. |
| 11:22.54 | jwpierce3 | then you can drop username/password |
| 11:22.57 | Lope | But I can email them. |
| 11:29.24 | Haris | is there a basic config for asterisk for doing basic video calling |
| 11:31.06 | *** join/#asterisk wil_syd (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 11:36.18 | jwpierce3 | Lope, your call is being rejected, probably due to authenication issue |
| 11:47.30 | Lope | yeah. |
| 11:47.30 | *** join/#asterisk acidfu_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
| 11:52.29 | Lope | cool I got an email back from them with an example asterisk config. |
| 11:57.31 | jwpierce3 | Lope, are you just going to be using one phome? |
| 11:57.33 | jwpierce3 | phone |
| 11:59.02 | Lope | I've got a few softphones, then I've got (at the moment) 1 VoIP provider. |
| 11:59.12 | Lope | Their sample config is also not working. Same problem. |
| 11:59.36 | Lope | what's very curious is as I said before. it says that my softphone failed the invite. |
| 12:00.13 | Lope | But the reply comes back from the VoIP provider which is a different account, the username is a DID. |
| 12:01.03 | Lope | there's something weird about me dialing with the context of that softphone. |
| 12:01.39 | Lope | Perhaps I need to change some channel variable before I can dial on a different SIP account? |
| 12:03.21 | Lope | YES!!! |
| 12:03.22 | Lope | working |
| 12:03.31 | Lope | ah, quality is rad. |
| 12:04.51 | Lope | so the problem is I was using Dial(SIP/1026@my.voipprovider.com,30) but it should have been Dial(SIP/1026@5555555555,30) where 5555555555 is my DID from my VoIP provider. |
| 12:13.36 | Lope | What is the period (dot) after all the XXXX? exten => _XXXXXXXXX.,n,Dial(SIP/${EXTEN}@27870010000) |
| 12:14.41 | Lope | oh, 1 or more characters |
| 12:15.45 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 12:16.13 | Lope | I'm trying exten => _021XXXXXXX,n,Dial and my number I'm dialling is 0211234567. Asterisk says it's not found in the dialplan. |
| 12:18.36 | Lope | oops, I didn't start the dialplan with 1 |
| 12:22.45 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:6ef0:49ff:fe50:659d) |
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| 13:00.27 | Haris | can asterisk handle wss:// format ? |
| 13:00.44 | Haris | web service socket ? |
| 13:01.44 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
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| 14:00.38 | hexanol | someone is aware of a memory leak in asterisk that appeared between asterisk 13.7.2 and asterisk 13.9.1 ? |
| 14:01.08 | hexanol | i.e. that got introduced in either in asterisk 13.8 or asterisk 13.9 |
| 14:01.23 | file | if it's not in JIRA, then no |
| 14:09.15 | Rasputin3711 | Can anybody work with yeastar? |
| 14:10.10 | [TK]D-Fender | We have no idea what their hiring practices are like. |
| 14:10.28 | [TK]D-Fender | They could be discriminatory, go find out... |
| 14:34.50 | *** join/#asterisk swimmercol (~lancelot@4E9D6FC8.rev.sefiber.dk) |
| 14:36.50 | scv | dialplans yay |
| 14:44.37 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:6355:b61:14e7:dbba) |
| 14:48.16 | tompaw | Is there a way to debug a mixing bridge self-optimization? i.e. to confirm that when there are only two participants it's doing things the most efficient way? |
| 14:48.27 | tompaw | Also, morning. |
| 14:48.41 | WIMPy | core show channels |
| 14:51.51 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-13.wall.uls.co.za) |
| 14:52.05 | tompaw | hmmm |
| 14:52.07 | tompaw | http://hastebin.com/ijeguwanup.mel |
| 14:52.19 | tompaw | why isn't the Snoop/ channel listed as a bridge participant? |
| 14:52.55 | tompaw | is it because it can't talk? |
| 14:53.26 | WIMPy | internals... |
| 14:54.11 | tompaw | WIMPy: are you saying it's by design? |
| 14:55.09 | WIMPy | yesUnot that i've ever seen a "Snoop/" anyway. |
| 14:55.31 | WIMPy | But I may never have looked at the right time. |
| 14:55.33 | scv | the dogg's favorite channel |
| 14:55.50 | tompaw | if you look at my pastebin, Snoop/1464619267 is definitely snooping on cid-512830, and yet it's not listed. |
| 14:56.04 | file | is the Snoop channel in the bridge? |
| 14:56.29 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:57.05 | tompaw | yes |
| 14:57.12 | file | did you add it to the bridge? |
| 14:58.08 | tompaw | yes, it works as intended |
| 14:58.12 | tompaw | i.e. it's recording the bridge. |
| 14:58.43 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 14:58.44 | file | it's snooping on a specific channel - that is it's getting the media you've requested from that channel and directing it elsewhere |
| 14:58.53 | file | that, however, does not mean that Snoop channel is in the bridge |
| 14:59.02 | file | the channel it is snooping on is in it |
| 14:59.06 | tompaw | ooh, I see, you might be right. |
| 14:59.09 | tompaw | let me confirm that |
| 15:00.17 | file | if you put it in the bridge then the bridge would get another copy of all the media in it, over, and over, and over |
| 15:00.27 | tompaw | boy do I feel stupid. |
| 15:00.31 | tompaw | you're right. |
| 15:01.06 | tompaw | so back to my original question, if it says Technology: simple_bridge, does it mean it's a simple 2-way no-transcoding dialogue? |
| 15:01.26 | file | it may or may not be transcoding, you'd have to check each individual channel |
| 15:01.40 | file | simple is just two channels passing frames back and forth |
| 15:02.17 | file | WIMPy, and a Snoop channel is an ARI thing - it's the method for injecting media into/listening on channels |
| 15:02.59 | tompaw | ha, got it, I think I know why the performance of my boxes is so crappy (10-15 bridges max per DL160) |
| 15:03.31 | scv | what generation DL160 |
| 15:03.37 | scv | or better, what CPU |
| 15:03.53 | WIMPy | Ok, buit as you just explained still the same as any other way of spying on a channel. |
| 15:03.54 | tompaw | while my Agent's channels show No transcode, the PSTN ones are listed as: http://hastebin.com/qeviyexifu.coffee |
| 15:04.26 | tompaw | WIMPy: file is 100% right, it attaches to one of the bridges' participating channels, effectively recording the bridge :-) |
| 15:04.36 | file | WIMPy, yeah the difference just being that the interface presents you a channel that you can do any channel stuff on - so to implement MixMonitor you snoop on a channel, and then you can record that Snoop channel |
| 15:05.22 | tompaw | so at the end of the day, here's what I'm getting: http://hastebin.com/uhamofibeq.coffee |
| 15:05.41 | WIMPy | A channel type for doing the snooping/spying, understood. |
| 15:06.05 | tompaw | scv: latest gen9 |
| 15:06.09 | tompaw | 1.6 Ghz |
| 15:06.13 | scv | and all you get is 10-15 bridges? o.O |
| 15:06.15 | file | it's the slin192, that's something gtjoseph has been looking into |
| 15:06.27 | tompaw | Intel(R) Xeon(R) CPU E5-2603 v3 @ 1.60GHz |
| 15:06.40 | scv | you must have something else eating a lot of resources |
| 15:06.43 | file | not sure why it'd do that though |
| 15:06.44 | tompaw | scv: yep, I've been scratching my head because of this for a really long time now. |
| 15:06.46 | scv | i'm still using much much older gear and have higher density |
| 15:07.08 | file | tompaw, what's the formats on the Snoop? |
| 15:07.12 | tompaw | scv: there's nothing else there, these are diskless servers, asterisk only, bridges + ari + snooping to an nsf directory |
| 15:07.17 | tompaw | file: wav |
| 15:07.19 | scv | tompaw: i'm using dual L5240 for my PBX nodes and have ~100 separate asterisk instances running on each machine |
| 15:07.24 | file | I mean the formats in core show channel |
| 15:07.27 | scv | so there must be something else causing your resource consumption |
| 15:07.28 | tompaw | oh sorry |
| 15:07.42 | jrun | missing logger.so, what happneded here? |
| 15:08.04 | jrun | this is fresh install from 13.9.1 |
| 15:08.13 | tompaw | file: http://hastebin.com/ikosusifig.coffee |
| 15:08.50 | tompaw | scv: we tried 3 different OS'es, 2.x, 3.x and 4.x kernels, different servers, same story. |
| 15:08.50 | gtjoseph | yep, you're hititng the slin192 transcoding issue. |
| 15:09.17 | jrun | 'logger reload' seems to parse logger.conf but 'module load logger' fails with: Error loading module 'logger': not found under /usr/lib/asterisk/logger.so |
| 15:09.17 | tompaw | wow, I feel like I'm finally getting somewhere |
| 15:09.18 | scv | tompaw: really odd |
| 15:09.26 | tompaw | gtjoseph: might that be the cause of my cpu overload |
| 15:09.43 | file | yeah, something is pegging it at slin192... the Snoop code should determine it to be slin @ 8000, dunno why it'd go for 192... |
| 15:09.46 | gtjoseph | i'm reaading the chat log... |
| 15:10.14 | tompaw | gtjoseph: long story short: 6c E5-2603 gives me around 10-15 bridges / server of sustainable load |
| 15:10.22 | tompaw | asterisk + ari + snooping, nothing else. |
| 15:10.23 | file | tompaw, how was the PSTN leg created? |
| 15:10.34 | file | is it a Local channel? |
| 15:11.20 | file | ah, you have some Local channels in use... yeah... |
| 15:11.50 | file | https://issues.asterisk.org/jira/browse/ASTERISK-26070 |
| 15:12.33 | tompaw | file: it calls Local/xxx@cont-fas and bridges it into the cid-xxx via ARI |
| 15:12.38 | tompaw | dialplan: http://hastebin.com/saqafatazi.coffee |
| 15:12.59 | tompaw | it's pretty straightforward, FAS (so the agent can hear the ringtone), caller id, call. nothing fancy. |
| 15:13.03 | gtjoseph | Yep, any time you create a local channel with ARI you can hit the issue. |
| 15:13.14 | tompaw | say whaaaaa? |
| 15:13.23 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:13.48 | gtjoseph | ... let me clarify... |
| 15:14.09 | gtjoseph | if you create a local channel and don't specify an "originator". |
| 15:14.46 | gtjoseph | when that happens, the ari code doesn't know how to constrain the formats a local channel can use so it adds all audio formats. |
| 15:15.31 | gtjoseph | when the time comes to make two channels compatible, the best format is used to preserve quality and in this case is slin192. |
| 15:15.48 | tompaw | omg |
| 15:15.58 | tompaw | yes, this is exactly what I'm doing |
| 15:16.06 | tompaw | http://hastebin.com/oqadozohuq.coffee << added .originate python code |
| 15:16.42 | gtjoseph | Yep, that's it. |
| 15:17.35 | gtjoseph | Can you specify originator channel on 1 of the Local's and see what happens? |
| 15:18.03 | tompaw | yes, let me just read the manual, tbh I'm not familiar with the concept of "originator" |
| 15:18.23 | gtjoseph | it's just the opposite channel |
| 15:18.44 | tompaw | right, but if they're BOTH local? |
| 15:18.51 | gtjoseph | you'd have to create 1 first without originator, then create the other with the originator specified as the forst one. |
| 15:19.00 | tompaw | ic. |
| 15:19.14 | tompaw | let me try this now, will take me 10-15 mins as I need to pull one server from the production |
| 15:19.20 | tompaw | if this works... damn. |
| 15:19.35 | gtjoseph | I actually haven't tested to see3 if this will work for 2 local channels though. |
| 15:20.24 | gtjoseph | also is it possible to convert the local channel that goes to the pstn to a specific technology channel? PJSIP/... |
| 15:21.18 | gtjoseph | maybe do the callerid manipulation in python instead of the dialplan context? |
| 15:22.20 | gtjoseph | If that doesn't work, you can also try the patch at https://gerrit.asterisk.org/2917 |
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| 15:25.03 | gtjoseph | jrun: logger isn't a separate module. it's part of the main asterisk binary. |
| 15:28.11 | tompaw | gtjoseph: it would be easier to convert the Agent channel, as it's literally just one line: http://hastebin.com/usoqetated.coffee |
| 15:28.17 | tompaw | plus it's called first anyway. |
| 15:28.31 | gtjoseph | perfect |
| 15:28.54 | tompaw | hm... I wonder what the reason was behind setting it up as Local/ in the first place. |
| 15:30.26 | tompaw | I know, I wanted to use PJSIP_DIAL_CONTACTS |
| 15:30.46 | gtjoseph | yeah, i just saw that. |
| 15:30.52 | tompaw | as these guys LOVE to roam between different stations for no particular reason. |
| 15:30.57 | gtjoseph | do the agents normally have more than 1? |
| 15:31.23 | tompaw | if this works, I'll chain them myself. |
| 15:32.50 | tompaw | crap, there's another reason, too. when I tear down the bridge, I scan the context of its channels to determine if there are any pstns or agents left in it and act accordingly. |
| 15:33.14 | tompaw | gtjoseph: how about I try the originator with 2 Locals first, see if that helps. |
| 15:33.22 | gtjoseph | sure! |
| 15:33.54 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:41.32 | tompaw | gtjoseph: is it the x.id or x.json['name'] I should pass as the originator? |
| 15:41.37 | tompaw | (x being the prior channel) |
| 15:41.41 | gtjoseph | id |
| 15:41.59 | tompaw | like '1464622838.61108' |
| 15:42.01 | tompaw | ok |
| 15:42.12 | gtjoseph | let me double check |
| 15:42.58 | gtjoseph | i just did this Thursday and I forgot already. :) |
| 15:43.17 | tompaw | how did you manage to find out what causes this issue? |
| 15:43.29 | gtjoseph | it's the "name". |
| 15:43.49 | gtjoseph | was doing a stress test and happened to be using alocal channel |
| 15:44.07 | tompaw | ok, so the full Local/blablabalb;1 ? |
| 15:44.37 | gtjoseph | i'm looking |
| 15:44.43 | tompaw | by the way, what do you use for your ARI stuff, python by any chance? |
| 15:45.07 | gtjoseph | it's actually "channelId" |
| 15:45.32 | gtjoseph | the Asterisk testsuite is all python |
| 15:45.50 | gtjoseph | if you want some ideas, nose around there. |
| 15:46.23 | tompaw | I don't understand how ARI is not super popular yet. I see it as the biggest thing in digital telephony of the past few years. |
| 15:46.36 | gtjoseph | there are a bunch of classes in the ./lib/python/asterisk directory. |
| 15:46.56 | gtjoseph | that's because developers are poor marketers. :) |
| 15:47.07 | file | well, and users don't... say they are using it |
| 15:47.29 | tompaw | in a month's time or so I will be able to provide a nice use case if you're interested |
| 15:47.42 | gtjoseph | absolutely! |
| 15:48.13 | tompaw | we have a very nice call center in the business district of panama, with 60 TVs all around the floor showing live stats, QA agents snooping on calls wirelessly, etc. |
| 15:48.51 | tompaw | and if this originator works, we won't need the nuclear reactor anymore! |
| 15:49.21 | gtjoseph | Ha! |
| 15:49.59 | file | there's a distributed call center I know about using ARI + WebRTC, the only reason I know is because I'm friends with the person on Facebook and he posted about it - cause it just worked for them >_> |
| 15:50.12 | tompaw | we settled on WebZoiper |
| 15:50.41 | tompaw | but our dialer is fully integrated with our bespoke crm, with kerberos SSO, etc. |
| 15:50.55 | tompaw | I really wish Zoiper didn't require windows tho :/ |
| 15:52.23 | tompaw | WriteTranscode: No, ReadTranscode: No |
| 15:52.28 | tompaw | Boom, baby! |
| 15:52.53 | gtjoseph | Yay! |
| 15:54.33 | tompaw | OK, now let's try on a production server with recording. |
| 16:02.58 | tompaw | works fine, pushed live, let's test it with 150 agents :-) |
| 16:04.36 | tompaw | I really hope that was the bit hogging up my CPUs... |
| 16:05.41 | gtjoseph | good news. let us know. |
| 16:06.16 | tompaw | I will have the results in 15-30 mins |
| 16:14.07 | tompaw | It's weird how a single 2-party call (+a snoop) spawns 7 channels. |
| 16:14.24 | scv | is there a perl ari library |
| 16:15.16 | scv | suppose i could just write the library myself |
| 16:15.59 | tompaw | yeah it's not too tricky, rest + websockets and you're all set. |
| 16:18.09 | gtjoseph | There's python and nodejs code at https://github.com/asterisk but no perl, sorry. as tompaw said though, just it's rest + websockets. |
| 16:19.24 | scv | simple enough then |
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| 16:26.25 | tompaw | ok, my celebration was a bit premature |
| 16:26.36 | gtjoseph | hmmm. |
| 16:26.45 | tompaw | the Local/ channels are not showing transcoding now, but their respective PJSIP channels do. both legs. |
| 16:27.01 | gtjoseph | yeah i was afraid of that. |
| 16:27.34 | gtjoseph | you're still just setting originator correct? you didn't change one of the locals to pjsip? |
| 16:28.00 | tompaw | yes, but it's kinda pointless if the referencing channel's PJSIP is also slin192 |
| 16:28.44 | gtjoseph | that is a little weird. what are the codecs set in the pjsip endpoints? |
| 16:28.50 | tompaw | ulaw |
| 16:29.01 | tompaw | and they're both showing (slin@192000)->(slin@8000)->(ulaw@8000) |
| 16:29.12 | gtjoseph | do you also have a !all or a deny=all? |
| 16:29.37 | tompaw | disallow=all |
| 16:29.37 | tompaw | allow=ulaw |
| 16:29.44 | gtjoseph | ok. |
| 16:30.02 | gtjoseph | try the local to pjsip conversion. |
| 16:30.06 | tompaw | I guess I'll have to try and call the agent without local |
| 16:30.08 | tompaw | yep |
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| 16:33.31 | tompaw | ah ffs, I left alaw,ulaw in my flowroute configuration and guess which one was picked up :> |
| 16:33.45 | gtjoseph | :) |
| 16:34.23 | file | achoo. |
| 16:35.31 | tompaw | ok, now it works :-) |
| 16:36.19 | tompaw | but I need to rewrite another piece of code that relied on this Local context to recognize agents. |
| 16:37.05 | gtjoseph | you could try the gerrit patch i referenced. with it, you can specify the format on the local channels. "format=ulaw" |
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| 16:37.17 | jrun | looking for eye-candy module-tree dependency view; any idea? |
| 16:37.33 | jrun | that is for runtime. |
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| 16:38.47 | tompaw | gtjoseph: I'll wait for it to hit master, don't want to make things more difficult for my admins if I can handle it myself. |
| 16:39.06 | gtjoseph | makes sense. |
| 16:39.12 | tompaw | I supposed they will merge it at some point? It sounds like a reasonable feature. |
| 16:39.47 | gtjoseph | oh yeah. within a week or two i'd guess. |
| 16:39.50 | tompaw | It's not too much of a hassle for me, too. Everything that's "*Local*" in the bridge will now be PSTN, everything "PJSIP" would be an agent. Simples :-) |
| 16:40.31 | tompaw | then I guess in a week we'll rebuild our PXE image with this new version. |
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| 17:04.38 | tompaw | ok, now it does work. pushing to production. |
| 17:22.13 | tompaw | %Cpu(s): 0.4 us, 0.4 sy, 0.0 ni, 99.2 id, 0.0 wa, 0.0 hi, 0.0 si, 0.0 st |
| 17:22.14 | tompaw | :D |
| 17:22.24 | gtjoseph | nice! |
| 17:22.31 | file | yay |
| 17:22.54 | tompaw | I'm waiting for 80% of the floor to come back from their lunch break, but that was taken off an inbound server that was usually under 40-50% load |
| 17:22.58 | scv | congrats tompaw |
| 17:23.12 | tompaw | I really hope we'll get similar results on outbounds |
| 17:23.14 | tompaw | :-) |
| 17:24.00 | tompaw | I can now work on my redis-list-based holding bridges! |
| 17:26.39 | tompaw | I hope this will take away the segfaults... |
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| 17:40.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 17:41.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 17:44.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 18:17.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 18:56.34 | Lope | how can I do web requests? I've tried loading res_curl.so func_curl.so res_config_curl.so |
| 18:56.46 | Lope | but `core show application curl` is not finding anything. |
| 18:57.15 | Lope | I also tried loading res_http_post.so but there wasn't any http command either. |
| 18:59.38 | [TK]D-Fender | FUNCTION <- |
| 19:00.15 | Lope | TKD, ah, thanks. |
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| 19:04.08 | Lope | all the curl examples I've seen are doing stuff like this: Set(foo=${CURL(http://somewhere.com/somepage.html?x=5&y=4)}) |
| 19:04.34 | Lope | I know it sounds obvious, but can you put actual variables in as parameters? (after the equals signs) |
| 19:05.51 | [TK]D-Fender | It's a function like any other |
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| 19:06.52 | Lope | okay. Does it block until it has the full result? |
| 19:07.12 | Lope | some of the examples I've seen put a 5 second wait command after it. |
| 19:08.07 | [TK]D-Fender | The only thing that is not blocking is Background() |
| 19:08.57 | [TK]D-Fender | And that's assuming you aren't using any time-consuming functions in the Dta |
| 19:14.45 | Lope | okay thanks |
| 19:14.47 | Lope | night night |
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| 20:36.13 | tompaw | Great news: my load stays low, incredibly low! Bad news: that didn't fix the occasional segfaults. |
| 20:37.00 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:37.32 | file | what is a backtrace from the segfault? |
| 20:41.16 | tompaw | one sec. |
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| 20:53.33 | tompaw | file: http://hastebin.com/epozixiger.css - don't have a full trace yet, because these are diskless servers, but we're on it as we speak. |
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| 20:53.50 | file | that is useless to me, sadly |
| 20:56.34 | tompaw | I know, we're trying to change the configuration to collect a full trace. |
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| 20:57.16 | WIMPy | crashed app_queue twice yesterday. The channel name was big pile of random data. |
| 20:58.02 | WIMPy | Have to try if I might even be able to reproduce that. It happened twice directly after a fresh start. |
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| 22:19.48 | tompaw | file: we've added NFS-based coredumps, will have some data for you tomorrow. |
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