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| 00:24.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 10:02.59 | madhu883 | Hello everyone, I have installed asterisk 13 in vm ware and connected asterisk to Matrix SETU ATA211G hardware device through IP address. I'm trying to initiating outbound call but asterisk respond with ERROR: pbx_spool.c:413 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 10:03.17 | madhu883 | <PROTECTED> |
| 10:03.45 | madhu883 | anyone please help me |
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| 13:05.03 | danielyk | @file: It is able to set "Require: mediasec" for REGISTER? |
| 13:05.19 | file | if you modify the code... |
| 13:07.51 | danielyk | Okay, then I have to modify the code... Not an easy thing with this complexity... -.- |
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| 13:30.43 | Kobaz | i would take a buyout from 3cx if it was big enough |
| 13:30.58 | Kobaz | oh whoops |
| 13:31.05 | Kobaz | i was replying to a message from like a week ago |
| 13:31.09 | Kobaz | (forgot to scroll down) |
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| 15:45.19 | perlmongler | Hello team, Have a PBX with about 60 phones across a city all connected through ASA tunnels and metro ethernet. Things work fine for a while but over a day or two phones will take 30 seconds to intiate a call with asterisk. I can see it dial ofter i hangup some times |
| 15:45.52 | perlmongler | My question is, how do I debug this and optomize this? The issues goes away after restarting the asterisk core |
| 15:46.15 | perlmongler | System resources, call volume, threads are all very low |
| 15:46.23 | perlmongler | doesn't look like a system resource issue |
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| 16:08.29 | Samot | What is an ASA tunnel? |
| 16:08.38 | Samot | You mean tunnels setup on a Cisco ASA? |
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| 16:16.01 | perlmongler | Yes |
| 16:16.30 | perlmongler | Cisco ASA VPNs, hardware |
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| 16:17.04 | perlmongler | Turned off SIP Inspection and that helped but didnt eliminate the issue, seems like it just takes longer to end up in the same nearly unresposive state |
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| 16:29.23 | Samot | OK so this sounds like an issue with how the VPNs are setup |
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| 16:34.28 | perlmongler | I agree but I am not sure what, because it works wounderfuly and slowly degrades |
| 16:34.46 | perlmongler | Woundering if anyone had ideas of settings to look at |
| 16:34.49 | perlmongler | timeouts? |
| 16:35.07 | perlmongler | or will none of the asterisk / phone settings help |
| 16:41.53 | drmessano | perlmongler: Are the tunnels set up on-demand? |
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| 16:50.15 | perlmongler | No, 1 to 1 site tunnels |
| 16:52.03 | perlmongler | @drmessano |
| 16:52.12 | perlmongler | Oh amn i am so used to slack now |
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| 17:19.22 | drmessano | That wasnât what I asked |
| 17:19.47 | drmessano | You can set up tunnel to only establish when theyâre needed to pass traffic |
| 17:19.51 | drmessano | Tunnels |
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| 17:23.10 | drmessano | Is there other traffic on these tunnels? Almost sounds like they're going down over time and constantly need to come back up when a call is placed |
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| 17:40.16 | dan_j | Hi. Looking at my logs, I just noticed that StopMixMonitor doesn't seem to be completing before the dialplan continues executing. My logs show the StopMixMonitor command plus the next two lines of the dialplan before "app_mixmonitor.c: End MixMonitor Recording" is shown. |
| 17:40.58 | dan_j | Is it expected that StopMixMonitor exits before Asterisk has actually stopped the recording? |
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| 17:43.37 | file | it is asynchronous |
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| 17:58.20 | sibiria | dan_j: if it were synchronous it wouldn't be able to record anything happening from your end |
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| 18:39.46 | exuberocity | Got disconnected - I'll re-ask |
| 18:39.48 | exuberocity | Hey guys, trying to find a log file of my SSH failed attempts -- FreePBX/Asterisk. Not having any luck, which file should I be looking in? |
| 18:45.20 | sibiria | i think that's more of a #freepbx question |
| 18:46.18 | sibiria | asterisk itself doesn't involve itself with being an SSH server |
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| 19:00.10 | dan_j | file: is there any way to confirm that it's closed the file before executing an external command? eg. A callback given to StopMixMonitor. |
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| 19:07.51 | sibiria | StopMixMonitor won't continue until it has closed the file |
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| 19:25.17 | dan_j | sibiria: That's not what file said and not what I've seen in production |
| 19:25.36 | file | I know of nothing off the top of my head |
| 19:26.43 | sibiria | dan_j: MixMonitor() is asynchronous, which it obviously must be in order to record both Us and Them |
| 19:27.09 | sibiria | StopMixMonitor() is not asynchronous because it doesn't need to, and shouldn't. it stops recording and frees the filehandle, then continues |
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| 19:27.35 | file | it doesn't |
| 19:27.42 | file | it tells the thread that is doing the recording and monitoring to stop |
| 19:28.15 | file | and then carries on |
| 19:32.37 | sibiria | since what version of asterisk? |
| 19:33.07 | sibiria | in asterisk 13 we have the problem with our dial plans getting stuck in various places with too high disk i/o during high call volumes, StopMixMonitor() being one of the calls that can take unusually long to finish |
| 19:33.43 | file | since... 11? or earlier |
| 19:33.55 | file | if the MixMonitor thread is blocked, then StopMixMonitor would be blocked |
| 19:37.07 | dan_j | I run StopMixMonitor in a Hangup handler. Are there any performance issues or blocking that I need to consider, before I add Wait(1) on the line after StopMixMonitor in the handler? |
| 19:41.52 | Samot | sibiria: Define "high call volumes" |
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| 19:47.22 | sibiria | Samot: up towards 1500 simultaneous calls |
| 19:48.10 | Samot | And how big is the server? |
| 19:48.13 | Samot | CPU/RAM? |
| 19:49.38 | sibiria | 8 cores, 32gb of ram. we're not seeing any processor or memory problem on behalf of asterisk itself. it's the rather slow disk i/o being the problem (EC2 instances with EBS volumes currently) |
| 19:50.14 | Samot | OH |
| 19:50.15 | Samot | JFC |
| 19:50.21 | Samot | Get off of AWS. |
| 19:50.49 | sibiria | no can do. clever corporate decision to move off of own hw |
| 19:51.02 | sibiria | also because they insist on running aws aurora for db needs |
| 19:51.28 | Samot | OK |
| 19:51.36 | Samot | Then this is going to be typical. |
| 19:51.42 | sibiria | which has turned out to contrive the gruesome trick of being both the shittiest, crappiest yet most expensive rdbms solution available |
| 19:51.55 | Samot | Tell the clever corp types, this is SOP from now on. |
| 19:52.26 | sibiria | we know it's not an asterisk problem, really. i'm sure it'll play just fine on direct SSD instead of EBS storage |
| 19:52.37 | Samot | Sure |
| 19:52.49 | Samot | Pay more for something you could be paying about 40% less at other places. |
| 19:52.57 | sibiria | AWS in a nutshell |
| 19:52.59 | sibiria | overpriced shit |
| 19:53.11 | Samot | It's horrible for high volume voip |
| 19:53.17 | Samot | You're behind NAT |
| 19:53.22 | Samot | The EBS is shitty as hell |
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| 20:28.57 | Kobaz | speaking of servers |
| 20:29.10 | Kobaz | anyone know a decent VPS provider with Central-US physical locations? |
| 20:34.27 | pa | hi |
| 20:34.38 | pa | is any1 having issues with Motif/google voice? |
| 20:35.22 | pa | i started to get "Everyone is busy/congested at this time (1:0/0/1)" when routing calls through google voice |
| 20:35.30 | pa | i cant figure out why |
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| 20:37.50 | file | it's dead. |
| 20:37.51 | file | gone. |
| 20:37.52 | file | kaput. |
| 20:37.59 | file | you can't use Motif with Google Voice. |
| 20:38.39 | file | there is a community patch to PJSIP to add support for connecting using SIP, https://github.com/naf419/asterisk/tree/gvsip - it has not yet been included in Asterisk and as always, your mileage may vary and it could stop working at any time |
| 20:39.30 | pa | file: since when? |
| 20:39.33 | Kobaz | * as per tradition on any google product |
| 20:39.37 | pa | it worked for me till last week or so |
| 20:39.40 | file | last week I think it finally died |
| 20:39.43 | pa | aha |
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| 20:40.21 | pa | bloody google :-) |
| 20:40.52 | pa | thanks for the link |
| 20:40.59 | pa | i'll see if i can patch my old asterisk 11 |
| 20:41.04 | drmessano | You can't |
| 20:41.43 | file | Asterisk 11 does not have PJSIP. |
| 20:42.39 | drmessano | Also, even if you do get it running, it's not a public API |
| 20:42.50 | drmessano | or public interface, I guess |
| 20:43.46 | drmessano | I just wonder how fast Google is going to tighten the screws once all these Asterisk boxes start using an interface clearly set up for a partner |
| 20:43.47 | pa | sad because it was cheap |
| 20:44.00 | pa | but maybe it's time to look for an alternative |
| 20:44.01 | drmessano | Get a real ITSP.. they're cheap too |
| 20:44.12 | drmessano | Flowroute, VoIP.ms, Twilio |
| 20:44.26 | pa | well 3c/min to mobiles worldwide was hard to beat |
| 20:44.32 | file | drmessano: they could kill it pretty easily by having the provisioning process issue a client certificate, and then validate it on the Google side |
| 20:44.33 | pa | never seen anything like that |
| 20:44.59 | file | so that without the TLS cert you couldn't even establish a TLS connection for SIP |
| 20:45.03 | file | (that'd be what I would do) |
| 20:45.18 | drmessano | Yeah.. I can't imagine it's gonna stay open like this for long |
| 20:45.37 | drmessano | If I was obihai, I would request it |
| 20:46.09 | pa | drmessano: indeed not bad.. |
| 20:46.14 | pa | (those you linked) |
| 20:46.19 | file | I use Telnyx myself |
| 20:46.38 | drmessano | Surely Obihai can ration some revenue loss from not selling Obi boxes for GV |
| 20:46.49 | Samot | By Obihai you mean Polycom. |
| 20:46.54 | pa | the first one seems to have the same rates as gv |
| 20:46.57 | pa | i mean flowroute |
| 20:47.04 | pa | is it a good provider? |
| 20:47.10 | Samot | Polycom bought Obihai this year. |
| 20:47.19 | drmessano | pa: No, I only recommended bad providers |
| 20:47.22 | Samot | Shortly after that, XMPP dies. |
| 20:47.31 | Samot | And OBITalks have GV branding... |
| 20:47.32 | pa | haha |
| 20:48.04 | Samot | I'm going to guess Polycom bought Obihai for the GV relationship. |
| 20:48.10 | drmessano | Anyway, as I was sayin |
| 20:48.18 | Samot | Or at least it was part of the reasons. |
| 20:48.20 | drmessano | It won't stay open like this for long |
| 20:48.26 | Samot | No way. |
| 20:48.29 | pa | sorry i meant voip.ms |
| 20:48.57 | drmessano | pa: I recommended 3 good providers, which is the point of making a recommendation |
| 20:49.59 | pa | :-) |
| 20:50.00 | pa | thanks for that |
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| 22:02.49 | Kobaz | anyone familiar with converting rj22 to 2.5mm |
| 22:07.22 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:21.04 | Ellenor | No |
| 22:23.58 | *** join/#asterisk stux|work (stux@cosmo.lunarshells.com) |
| 22:24.03 | Kobaz | i have this adapter but it produces massive echo on the far end |
| 22:26.55 | *** join/#asterisk gruetzkopf (gruetzkopf@bnc.dont-follow.us) |
| 22:31.06 | Ellenor | don't use it then |
| 22:31.32 | Kobaz | well i need a way to convert the 2.5mm jack to rj22 |
| 22:41.44 | Samot | Have you tried another adapter? |
| 22:41.51 | Samot | Or changed settings on the phone? |
| 22:42.12 | Samot | And what "far end"? |
| 22:43.01 | Samot | You mean the other party hears echo in your voice audio? |
| 22:43.53 | *** join/#asterisk sibyakin (~sibyakin@83.234.69.145) |
| 22:47.14 | Kobaz | i've tried rewriting the adapter |
| 22:47.21 | Kobaz | rewiring, er |
| 22:47.40 | Kobaz | i think there's a bridge in the female barrel end |
| 22:47.49 | Kobaz | i tried turning on echo can on the phone |
| 22:48.02 | Kobaz | a straight rj22 headset works finr |
| 22:48.14 | Kobaz | damnit, can't type for nuttin for nobody |
| 22:48.53 | Samot | Why did you try to re-wire the adapter? |
| 22:48.56 | Kobaz | yeah the other end of the call (the far end) the 'not myself' part of the call, echos back on everything the far end puts in |
| 22:49.08 | Kobaz | because everything is pointing to the adapter |
| 22:49.19 | Samot | OK so why didn't you get another one? |
| 22:49.22 | Samot | A different brand? |
| 22:49.32 | Kobaz | because i can't find other brands, so far |
| 22:49.33 | Samot | Or just a new one to rule out faulty equipment? |
| 22:49.46 | Samot | Did you try a different phone? |
| 22:49.52 | Kobaz | everything is coming up 'headset buddy' |
| 22:49.59 | Kobaz | everywhere i search, which is the problematic adapter |
| 22:50.05 | Kobaz | which apparently has a known issue with echo |
| 22:50.10 | Samot | OK. |
| 22:50.11 | Kobaz | ...according to the reviews |
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| 22:50.47 | Samot | Echo Cancellation / Suppression is hardware and/or software based |
| 22:50.55 | Kobaz | right |
| 22:51.02 | Samot | Messing with the wires isn't going to do much. |
| 22:51.09 | Kobaz | why wouldn't it |
| 22:51.13 | Kobaz | if there's an internal bridge |
| 22:51.20 | Samot | And it's software based? |
| 22:51.23 | Kobaz | rx/tx linked together, there's your problem |
| 22:51.33 | Samot | So did it fix it? |
| 22:51.36 | Kobaz | i didn't |
| 22:51.41 | Samot | Still there? |
| 22:51.42 | Kobaz | that's why i'm talking to you guys |
| 22:51.50 | Samot | What type of phone? |
| 22:51.53 | Kobaz | i have it on a breadboard and i'm trying various things |
| 22:52.04 | Kobaz | Polycom IP-550, IP-335, VVX 400 |
| 22:52.09 | Kobaz | all have the same problem with this adapter |
| 22:52.52 | Kobaz | phone A (with adapter + headset) calls phone B.... person on phone B says something,.. that person hears what they said back within a few ms |
| 22:53.01 | Samot | Our latest version of the 2.5mm headset to RJ9 adapter is finally here. Compatible with select Grandstream & Yealink phones as well as the Cisco 79xx and Avaya 96xx series phones. |
| 22:53.09 | Samot | ^^ Did you miss that part of the product details? |
| 22:53.20 | Kobaz | right |
| 22:53.24 | Samot | Or where they listed on Polycom's supported headset/adapter list? |
| 22:53.35 | Samot | So it's more like the polycom doesn't support it. |
| 22:53.36 | Kobaz | just because it doesn't list it specifically, doesn't mean it might not work |
| 22:53.40 | Samot | OK |
| 22:53.51 | Samot | But re-wiring it won't make it compatible. |
| 22:53.56 | Kobaz | why not? |
| 22:54.07 | Samot | Because it's the POLYCOM doing the stuff |
| 22:54.13 | Kobaz | it's just a mechanical connection |
| 22:54.22 | Samot | If it doesn't play nice with the adapter... |
| 22:54.32 | Kobaz | the right wires in the right locations, will make it work |
| 22:54.48 | Samot | No, it will not. |
| 22:54.48 | Kobaz | no different than plugging in a straight rj22 headset (with the wires in the right places) |
| 22:54.55 | Kobaz | i guess you don't do much electronics work |
| 22:55.21 | Samot | So how many different ways can you re-wire this? |
| 22:55.29 | Samot | To get the right wires in the right locations? |
| 22:55.34 | Kobaz | 4*3*2*1 |
| 22:56.33 | Kobaz | 24 isn't an unreasonable amount to go through, and some can obviously be eliminated because i've already found where TX goes |
| 22:56.41 | Samot | So you think that re-pinning the RJ22 port is going to do it? |
| 22:56.49 | Kobaz | possibly |
| 22:56.49 | Samot | Or end, which ever side this is... |
| 22:56.52 | Samot | No. |
| 22:57.02 | Kobaz | but i have a feeling, the problem is at the other end |
| 22:57.05 | Kobaz | the 2.5mm end |
| 22:57.12 | Kobaz | so i may need a new jack |
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| 23:18.24 | Kobaz | Samot: it's not the polycom... i figured it out... 2.5mm headsets are designed to tx/rx on the same pair |
| 23:18.52 | Kobaz | so, i was correct in my original hypothesis, is that the issue is on the 2.5 end |
| 23:18.56 | Kobaz | specifically the headset |
| 23:19.18 | Kobaz | at least i know for sure now |
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| 23:22.32 | Kobaz | and again, digressing to the original, right wires in the right locations... it would be totally not worth the effort, but splitting the mic and earpiece onto their own pairs would fix the issue |
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| 23:31.07 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-gnsdlinfijrtzqxo) |
| 23:35.23 | forgotmynick | i have asterisk running on a server behind nat. all trunks work fine but since adding sip2sip.info (plivo sends incoming calls to the SIP URI), asterisk keeps rejecting incoming calls on that number but shows our WAN IP address as the connection from: [2018-07-24 00:27:55] WARNING[7761][C-00000020] Ext. s: "Rejecting unknown SIP connection from XXX.XXX.XXX.XXX" - can someone point me in the right direction on how to |
| 23:35.23 | forgotmynick | debug this? |
| 23:36.26 | Samot | You have a trunk that matches the IP? |
| 23:36.42 | [TK]D-Fender | that is clearly not a SIP rejection |
| 23:36.50 | [TK]D-Fender | it is obviously hitting dialplan |
| 23:36.51 | Samot | And why are they sending the call to the s extension? |
| 23:37.31 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:39.46 | forgotmynick | good point |
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