| 13:13.33 | *** join/#asterisk infobot (ibot@rikers.org) |
| 13:13.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.12.0 (2016/10/25), 11.24.0 (2016/10/25), Standard: 14.1.0 (2016/10/25); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 13:14.04 | [TK]D-Fender | MY BITCH IS BACK! |
| 13:14.11 | [TK]D-Fender | ~botsnack |
| 13:14.11 | infobot | :), [TK]D-Fender |
| 13:18.15 | Samot | Nothing? |
| 13:18.16 | tparcina | When I try to play a file to a caller, it he hangs up fast, I get this warning message: app_playback.c:493 playback_exec: Playback failed on SIP/307-00000c63 for |
| 13:18.19 | Samot | Of course. |
| 13:18.49 | tparcina | Can I do anything to avoid those warnings in log? |
| 13:19.06 | Samot | Well it failed. |
| 13:19.26 | Samot | So don't call on audio files for playback that don't exist. |
| 13:20.27 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 13:24.37 | tparcina | Samot: That file exists. |
| 13:24.49 | Samot | Then show the call. |
| 13:24.56 | Samot | Show what's happening. |
| 13:24.57 | *** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com) |
| 13:27.05 | [TK]D-Fender | If it's complaining that it didn't get through the job... stop worrying about having that message in your logs. |
| 13:27.55 | [TK]D-Fender | * will log what it considers a failure. You clearly know better how to interpret it's meaning and actual impact, so simply stop worrying about it. |
| 13:28.28 | tparcina | Samot: http://pastebin.centos.org/56571/ |
| 13:28.31 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:29.29 | tparcina | [TK]D-Fender: OK, but it creates lots of false warnings in my log files, and it's hard to notice the relevant warnings. |
| 13:30.29 | [TK]D-Fender | then filter them out so you don't feel hurt by them |
| 13:30.57 | [TK]D-Fender | tparcina, Did you even look at what you PB'd there? |
| 13:31.16 | Samot | That's garbage. |
| 13:34.16 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 13:35.16 | tparcina | [TK]D-Fender: That's the call. |
| 13:35.53 | tparcina | Some DB data is entered and some files are played. |
| 13:36.15 | [TK]D-Fender | You left out even the ERROR message we are talking about. It shows NOTHING WRONG AT ALL |
| 13:36.20 | Samot | So there's no playback errrors.. |
| 13:36.33 | [TK]D-Fender | What value does that PB offer anyone? |
| 13:37.02 | [TK]D-Fender | Do we see a call drop? Do we see the message you were referring to? What's the point? |
| 13:42.07 | vlt | [TK]D-Fender: You descibed my problem very well :-) The fact that Originate should have to do nothing with audio is what it makes so difficult to debug. Weird behaviour that I could reproduce _only_ with Originate so far. |
| 13:42.28 | vlt | [TK]D-Fender: Any idea what could be misconfigured? |
| 13:42.37 | [TK]D-Fender | because how you set your peers up means there was ANOTHER side of the call involved on other things... |
| 13:42.47 | [TK]D-Fender | things liek REINVITES that might bypass having done * properly, etc |
| 13:43.06 | [TK]D-Fender | Obviously the situation is different than just talking with another device as dialed though * |
| 13:43.36 | [TK]D-Fender | Time to look at what your call is actually doing |
| 13:43.48 | vlt | [TK]D-Fender: How? |
| 13:43.54 | vlt | sip debug? |
| 13:43.56 | [TK]D-Fender | CLI, how else? |
| 13:43.57 | tparcina | Is it my fault that there is no error written there? |
| 13:44.00 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:44.00 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:44.03 | [TK]D-Fender | and yes, obvioously SIP debug. |
| 13:44.18 | tparcina | [TK]D-Fender: Check this form warning log: [2016-10-27 15:23:58] WARNING[28583][C-00000736]: app_playback.c:493 playback_exec: Playback failed on SIP/709-00000c7c for hrv/pukljucena |
| 13:44.29 | Samot | That wasn't in the call debug. |
| 13:44.37 | Samot | Show a debug with the error. |
| 13:44.50 | [TK]D-Fender | tparcina, You just said you were getting errors. You know that's what we're expecting to see. Why aren't you showing us a call where you are GETTING it? Why do you think we asked to see it in the first place? |
| 13:44.51 | *** join/#asterisk lordvadr (~lordvadr@unaffiliated/lordvadr) |
| 13:44.59 | *** join/#asterisk dandann00dle (uid120491@gateway/web/irccloud.com/x-bpmbtbzdvudboifm) |
| 13:45.18 | [TK]D-Fender | You were asked to SHOW the problem. that PB doesn't show it |
| 13:48.20 | tparcina | [TK]D-Fender: How should I collect data about this problem? |
| 13:48.34 | tparcina | [TK]D-Fender: Would sip debug be enough? |
| 13:49.15 | Samot | asterisk -rvvvvvvvvv |
| 13:49.20 | Samot | sip set debug on |
| 13:49.24 | Samot | replicate issue. |
| 13:49.42 | eric_hill | tparcina: Place a call into your system that plays back the file, and hang up on it to cause the error. |
| 13:49.50 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 13:53.23 | stefan27 | When asterisk has multiple chan_sip friends A and B registered at same IP port and asterisk receives an invite from that IP and PORT which peer is matched? |
| 13:54.11 | [TK]D-Fender | stefan27, friend = user+peer, and matches on USER first |
| 13:54.18 | [TK]D-Fender | 9From:) |
| 13:54.24 | [TK]D-Fender | (From:) |
| 13:55.17 | stefan27 | ok but if the From value is something boogus, does it pick an arbitrary user registered at that ip+port? |
| 13:55.42 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:55.54 | stefan27 | asterisk never uses the information in the contact header for matching? |
| 13:57.21 | Samot | stefan27: How are you getting two endpoints with the same IP and same port? |
| 13:57.24 | [TK]D-Fender | Not AFAIK |
| 13:58.12 | stefan27 | no idea; I'm just observing the input output behaviour of this asterisk (the asterisk is not ours) |
| 13:58.41 | Samot | What output behaviour? |
| 13:58.48 | Samot | Show these two endpoints. |
| 13:58.58 | stefan27 | I cannot, they contain customer data |
| 13:59.17 | Samot | asterisk -rx "sip show peer <peer>" |
| 13:59.37 | Samot | That's not going to contain anything sensitive about the account. |
| 13:59.48 | Samot | That's going to show the peer information in Asterisk. |
| 13:59.53 | Samot | No passwords... |
| 14:00.24 | stefan27 | Im not authorized, i'd have to mask every single field like ip and username |
| 14:00.35 | Samot | Then there's nothing we can do. |
| 14:00.45 | Samot | You are saying that the endpoints are registering with the same IP and port. |
| 14:00.47 | Samot | That's not possibe. |
| 14:00.51 | Samot | That's not possible. |
| 14:01.11 | Samot | So either you are reading the output completely wrong or there is an issue. |
| 14:01.31 | Samot | Unless you can show the things we request, we can't troubleshoot. |
| 14:01.54 | [TK]D-Fender | Officer : "Doctor we need an autopsy for cause of death for our investigation". Mortician : "Where's the body?". Officer : "WE CAN'T SHOW YOU THAT!" |
| 14:02.03 | stefan27 | Addr->IP is indeed the same for A and B |
| 14:02.09 | Samot | PROVE IT |
| 14:02.12 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 14:02.31 | Samot | If you are not authorized to share customer details.. |
| 14:02.40 | Samot | Don't go to a public IRC room asking for support. |
| 14:02.56 | tparcina | Samot: eric_hill: newtonr: OK, TY, I believe I have collected the data you need: http://pastebin.centos.org/56576/ |
| 14:03.41 | tparcina | I guess [TK]D-Fender was right. * just complains that he didn't managed to do what he was suppose to do. |
| 14:03.56 | Samot | More importantly, don't offer public utility services that require you to use IRC to support your customer base. |
| 14:04.09 | tparcina | But I don't think he should send warning because of this. |
| 14:05.05 | [TK]D-Fender | tparcina, It does, and can safely be ignored on its own ground. If you really dislike it then either filter your logs of those, or mod the code the generates them |
| 14:05.47 | infernix | I dial in from a specific cell number to my Asterisk server using a line on a SIP trunk, and through the DISA menu I dial an extension. This extension dials out to a number through that same SIP trunk. how can I get Asterisk out of the traffic path once the line has been picked up? |
| 14:05.54 | *** join/#asterisk johnkeates (~johnkeate@92.66.153.5) |
| 14:06.32 | stefan27 | I find your ideology very strange. This is a discussion forum; I can share parts of a use case which I find relevant; if you believe I'm leaving out relevant details you should ignore the answer, but there might be other users for which the information I provide is sufficient to guide me towards an answer |
| 14:06.46 | stefan27 | in IRC anyone can publish ideas and 0 or more people may answer |
| 14:07.09 | Samot | You have an issue with a customer right? |
| 14:07.10 | stefan27 | When I type a question or an idea here, somehow you view it as I open a support ticket to you personally |
| 14:07.14 | Samot | They would like it resolved? |
| 14:07.18 | Samot | You don't know what the issue is. |
| 14:07.20 | Samot | So you are here. |
| 14:07.37 | Samot | We don't know what the issue is completely. How you have things setup or how your enduser is setup.. |
| 14:07.42 | Samot | If you can't share ANY of those details... |
| 14:07.44 | Samot | It's pointless. |
| 14:07.50 | *** part/#asterisk insmodppa (~insmod@unaffiliated/insmodppa) |
| 14:08.31 | Samot | So yes, you can post vague questions and never actually show hard proof while just posting snippets and pieces of debugs.. |
| 14:09.39 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 14:09.54 | infernix | ah, never mind. I had a Dial(..,H) so it would never remotely bridge the call |
| 14:10.19 | Samot | infernix: ?? |
| 14:10.36 | Samot | You call Asterisk, Asterisk calls the number. |
| 14:10.45 | Samot | That's two channels that Asterisk bridges together. |
| 14:11.14 | infernix | Samot: if you have a call-in menu and use Set(_HANGUP=Hg) Asterisk will keep listening for DTMF sequences |
| 14:11.23 | infernix | so it can't remotely bridge |
| 14:11.24 | Samot | No, I got that..I see the H |
| 14:11.28 | Samot | Sorry. |
| 14:12.23 | infernix | i dropped the H and it started to do what i want :) |
| 14:17.19 | johnkeates | can I use an asterisk-asterisk combo to setup a fake sip trunk with real DID's but simulated in an internal network? |
| 14:17.33 | johnkeates | i think i can, but before i do something stupid, asking would probably not hurt :p |
| 14:17.34 | *** join/#asterisk kharwell (kharwell@nat/digium/x-aheywmpbajkytilk) |
| 14:17.34 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:17.39 | stefan27 | I have communicated with 1000s of people over IRC; I have never run across someone like you before Samot. Yes, in this channel I ask a lot more questions than I answer, but I never expect anyone to help unless they are interested. I dont think you are suited to set the rules for how to communicate here. |
| 14:18.05 | Samot | "Hi I need help with X" |
| 14:18.10 | Samot | "Hi, show us Y" |
| 14:18.12 | Samot | "I can't" |
| 14:18.19 | Samot | That's how this has gone. |
| 14:18.22 | johnkeates | XY problem detected |
| 14:18.24 | stefan27 | you're making a straw-man argument |
| 14:18.30 | Samot | OK. |
| 14:18.51 | Samot | So what is your theory how why they are showing the same IP and same port? |
| 14:19.36 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:19.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:20.28 | Samot | Wait.. |
| 14:20.39 | Samot | You said the Addr->IP is the same? |
| 14:20.53 | Samot | They are behind NAT, aren't they? |
| 14:22.28 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 14:23.07 | stefan27 | Many sip friends can be registered at same IP, and there's some mechanism in asterisk which upon receiving invite matches to sip peer based on sip package contents and ip src and ip port... all I'm trying to do is find the logic which controls this mechanism, [TK]D-Fender already pointed out sip-header From is relevant, but I'm trying to find out the full mechanism |
| 14:23.54 | Samot | They are behind NAT aren't they? |
| 14:24.06 | stefan27 | either by locating documentation somewhere or I'll have to dig into source code but it's not easy |
| 14:24.14 | stefan27 | I guess so |
| 14:24.36 | Samot | Either their endpoints are direct on the Internet or they are behind NAT. |
| 14:24.55 | Samot | If they are all coming from the same WAN IP, then they are behind NAT. |
| 14:25.14 | Samot | The Addr->IP field shows that information... |
| 14:25.22 | [TK]D-Fender | stefan27, the logic ... is the From: |
| 14:25.32 | Samot | The Reg. Contact field shows where the call will be sent. |
| 14:29.40 | tparcina | [TK]D-Fender: OK, TY |
| 14:30.30 | tparcina | Does it make a sense to contact developers and suggest that no warning should be created it * didn't managed to do something because the other side has hung up the phone? |
| 14:32.39 | [TK]D-Fender | Maybe some users want to know that it didn't finish. |
| 14:32.47 | [TK]D-Fender | And that is a surefire sign |
| 14:32.57 | [TK]D-Fender | Did you think of that? No, you only think about YOURSELF :p |
| 14:34.01 | [TK]D-Fender | It doesn't hurt you. It's just filterable words in a log. |
| 14:34.39 | tparcina | But the reason why it didn't finish should make difference. |
| 14:34.59 | igcewieling | Isn't that what HANGUPCAUSE is for? |
| 14:35.32 | tparcina | If * didn't do something because the file does not exist, go and write that warning. |
| 14:36.10 | tparcina | But it it normal that * can't play file to caller if he hangs up before it's played. And * shouldn't warn me because of that. |
| 14:36.19 | tparcina | But it is normal that * can't play file to caller if he hangs up before it's played. And * shouldn't warn me because of that. |
| 14:38.02 | [TK]D-Fender | <igcewieling> Isn't that what HANGUPCAUSE is for? <- doesnt' tell you that the playback didn't FINISH |
| 14:40.13 | [TK]D-Fender | IT'S A FUCKING WARNING STOP BEING A WHINY BITCH. NO, THERE IS ALREADY ANOTHER WARNING FOR "FILE NOT FOUND". THEY OBVIOUSLY MEAN *** DIFFERENT THINGS *** |
| 14:40.16 | [TK]D-Fender | <PROTECTED> |
| 14:42.20 | [TK]D-Fender | If you don't like the message and it bothers you that much, you've got the source.. go mod it to remove all reference to that warning and recompile |
| 14:45.04 | *** join/#asterisk matrix1233 (~matrix123@197.1.158.122) |
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| 14:47.06 | [TK]D-Fender | :) |
| 14:50.27 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 14:52.39 | johnkeates | do a sed on the sources, replace "error" with "oopsies :'(" |
| 14:52.55 | johnkeates | makes for a lot more fun software :p |
| 14:57.04 | [TK]D-Fender | "FATAL CRASH" shall henceforth be called "Special features!" |
| 14:57.04 | [TK]D-Fender | #rebranding |
| 14:57.15 | johnkeates | haha yeah |
| 14:58.01 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 14:58.55 | [TK]D-Fender | "Here at Acme Tech we get to name our salary. I named mine 'Fred'!" |
| 15:00.03 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 15:00.29 | johnkeates | I name mine 'Cheese'. Every time I go to pickup my pay check, they'll tell me "here is your cheese" |
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| 15:25.15 | mohan_ | hi i m very new to asterisk, i want to setup inbound with 2 call agents, can you please suggest how the dial plan should be |
| 15:26.30 | [TK]D-Fender | It should be however you want |
| 15:26.58 | [TK]D-Fender | You haven't described anything so how should anyone be able to suggest how to build it? |
| 15:28.10 | mohan_ | D-Fender: I want the calls to reach available call agents |
| 15:30.38 | mohan_ | our customers calling to a specific mobile no we provided earlier but that is not DID so we shall ask mobile vendor to do call forwarding to one of DID numbers |
| 15:31.21 | mohan_ | what should the dial plan be like |
| 15:35.15 | [TK]D-Fender | Depends |
| 15:35.37 | [TK]D-Fender | You're still talking about "agent" like that necessarily means something specific |
| 15:35.44 | [TK]D-Fender | Reach them HOW? |
| 15:35.48 | [TK]D-Fender | When? |
| 15:35.55 | [TK]D-Fender | How do you define "available"? |
| 15:36.25 | *** join/#asterisk monsterco (~monsterco@TOROON474AW-LP130-01-1177674931.dsl.bell.ca) |
| 15:37.07 | [TK]D-Fender | How do you define "reach?" Will it ring them forever? Sever different times? re-evaluate if they are "availabel" between possible attempts? |
| 15:37.18 | [TK]D-Fender | Should is say something before trying? |
| 15:37.26 | [TK]D-Fender | While trying? Should it stop at some point? |
| 15:37.48 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP130-01-1177674931.dsl.bell.ca) |
| 15:37.56 | [TK]D-Fender | Do something afterwards? |
| 15:38.24 | [TK]D-Fender | By this point you should realize you have not described ANYTHING. You are asking for steps to a flow you have not even put a description on. |
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| 16:11.44 | riess82 | good evening all. i have managed to set up a spa232d as gateway from a standard phone line to asterisknow. it works quite well, but if i reject a call, the softphone rings again, as if asterisk is handling the reject correctly |
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| 16:30.28 | *** join/#asterisk v1r0 (~viro@unaffiliated/viro) |
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| 17:03.16 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 17:03.21 | ModFather | Hi There, |
| 17:05.10 | ModFather | i have inside my queues.conf 4 different agents, and queue category = ringall .. the issue i have is that when an agent is already on a call every other incoming call rings on the softphone of the agent. |
| 17:07.24 | ModFather | [TK]D-Fender are you there? |
| 17:07.56 | [TK]D-Fender | yes, though you shouldn't expect personalized support for everything |
| 17:08.33 | [TK]D-Fender | Or delay your description and backup waiting for an acknowledgement |
| 17:08.50 | ModFather | I just asked you because i know that you are always helping, if you cant its not problem and sorry for disturb |
| 17:10.06 | [TK]D-Fender | just show what you have |
| 17:10.47 | ModFather | on a queue i have strategy=ringall when all agents are on a call, every time another call comes in, the new call rings in the ear of the agent on a call and disrupts them. is there a way we can have that call not ring in if they are on a call already? |
| 17:11.24 | [TK]D-Fender | just show what you have |
| 17:12.33 | ModFather | http://pastebin.ca/3733479 |
| 17:13.59 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 17:14.39 | [TK]D-Fender | And that is your WHOLE queues.conf ? |
| 17:15.23 | ModFather | yes |
| 17:16.01 | ModFather | i have setup a new fresh install on ubuntu , and asterisk 13.6.0 |
| 17:16.44 | ModFather | http://pastebin.ca/3733482 |
| 17:16.49 | ModFather | my whole extensions.conf |
| 17:16.54 | [TK]D-Fender | Go take a minute to read the sample config |
| 17:17.06 | ModFather | sure |
| 17:17.10 | ModFather | be right back. |
| 17:17.49 | [TK]D-Fender | Heck, spoil yourself. Take MORE than a minute. |
| 17:18.03 | ModFather | .. |
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| 17:24.07 | ModFather | ringinuse = no |
| 17:25.29 | [TK]D-Fender | It's almost as though that parameter was MADE for it.... |
| 17:26.08 | igcewieling | When I realized the Ubiquity enterprise management software didn't manage their own enterprise switch, I knew the system sucked. |
| 17:31.07 | ModFather | [TK]D-Fender i found related to my issue only the "ringinuse", i have tried it and it didnt worked |
| 17:34.34 | [TK]D-Fender | Show that you've configure it right, applied your changes and then the 2 calls to prove it |
| 17:34.43 | [TK]D-Fender | along with the status dump we never got |
| 17:37.55 | kfife | Does anyone know if there's a way to dynamically specify non-vm recording with MiniVMNotify? I can call System() but if I can use I'd like to send an email and inherit the template. |
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| 19:09.28 | Ecco | Hey |
| 19:09.43 | Ecco | I'm trying to use Asterisk to place outgoing calls using a SIP account |
| 19:10.00 | Ecco | I know this account is working (works on my softphone) |
| 19:10.19 | Ecco | I'm trying to configure it in Asterisk but quite don't know how to see if everything's ok |
| 19:10.20 | kfife | Anyone have experience using MyScribbit.com for voicemail transcription? |
| 19:10.29 | kfife | Or CISscription (not to discriminate) |
| 19:10.37 | igcewieling | Ecco: see .call files. |
| 19:10.47 | Ecco | Yeah I've looked at this |
| 19:10.57 | Ecco | but I don't even know what the Channel should be in the call file |
| 19:11.07 | igcewieling | Same as your normal Dial line usually. |
| 19:11.14 | Ecco | I thought I could type a few commands in "asterisk -r" to see if it could connect allright to the SIP account |
| 19:11.33 | igcewieling | why in the worrld would you think that? |
| 19:11.45 | Ecco | because I'm positive I forgot a setting somewhere |
| 19:11.50 | Ecco | There are just too many of them :) |
| 19:12.13 | kfife | Ecco: Yeah, there's literally dozens of options. |
| 19:12.19 | Ecco | indeed :) |
| 19:12.30 | Ecco | So I'd just like to diagnose if Asterisk could connect to my SIP account |
| 19:12.43 | kfife | You could literally have more than 12 different configuration with a power tool like Asterisk |
| 19:12.47 | Mango45 | Ecco: To validate your credentials, you need to place a call to the service provider. |
| 19:12.51 | Ecco | ok |
| 19:13.16 | igcewieling | What Dial line do you use for a normal phone to call using your SIP service? |
| 19:13.28 | kfife | Also, configure your softphoen and play yourself Playback(tt-monkeys) |
| 19:13.30 | Mango45 | You can also register, but you specifically said "outgoing call", so that may not work. |
| 19:14.16 | igcewieling | In case anyone is thinking of using Ubiquity UniFi hardphones -- they suck. |
| 19:14.35 | kfife | igcewieling : as I suspected. Just a gut feeling |
| 19:14.57 | Ecco | chan_sip.c: Received response: "Forbidden" from '"asterisk" <sip:....> |
| 19:15.08 | igcewieling | looks like you need to fix things. |
| 19:15.13 | Ecco | sure |
| 19:25.49 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 19:28.51 | Ecco | Duuuh |
| 19:28.58 | Ecco | asterisk[4243]: segfault at 484ec01c5f9d ip 00007fd33205b006 sp 00007fd3124ad290 error 4 in libsqlite.so.0.8.6 |
| 19:29.09 | *** join/#asterisk v1r0 (~viro@unaffiliated/viro) |
| 19:30.12 | igcewieling | did you build from source? |
| 19:30.16 | Ecco | no |
| 19:30.19 | Ecco | Plain Debian install |
| 20:02.42 | igcewieling | try building from source. Asterisk is almost the only thing I ever build from source. |
| 20:04.12 | Ecco | yay, it's working |
| 20:04.20 | Ecco | Now for some reason I cannot set the callerid |
| 20:04.39 | Ecco | No matter what I set it to, the original SIP account is displayed |
| 20:05.22 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 20:05.22 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 20:06.25 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 20:35.53 | ModFather | http://pastebin.ca/3733525 i have this queue config on my queues.conf, i have the following issue, when 1 from the phones are already in a call, the next call will ring too on the phone which is already in a call... how i can stop that? |
| 20:36.05 | ModFather | to prevent hear the ring from incoming call if i am already in a call? |
| 20:37.01 | *** topic/#asterisk by putnopvut -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.12.1 (2016/10/27), 11.24.1 (2016/10/27), Standard: 14.1.1 (2016/10/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 20:57.43 | *** join/#asterisk matrix1233 (~matrix123@197.1.158.122) |
| 20:58.58 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:59.49 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
| 21:01.39 | ModFather | [TK]D-Fender i have also try skipbusy yes still nothing :( |
| 21:02.02 | igcewieling | turn off call waiting on your phone |
| 21:02.43 | ModFather | igcewieling its the only way ? so it cant be from asterisk side? |
| 21:03.32 | [TK]D-Fender | I never got the backup I requested |
| 21:03.41 | igcewieling | ModFather: I'm sure there is a nice complicated way using Asterisk, but the easiest way is to disable it on your phone so your phone sends back a busy |
| 21:03.53 | [TK]D-Fender | No need for the phone to change |
| 21:04.00 | [TK]D-Fender | This should be an * decision |
| 21:04.07 | [TK]D-Fender | Time to see where he's screwing up |
| 21:04.23 | igcewieling | [TK]D-Fender: I presume by using a GUI |
| 21:04.44 | ModFather | [TK]D-Fender i saw you my configuration |
| 21:04.49 | ModFather | extensions.conf and queues.conf |
| 21:04.56 | *** join/#asterisk n3ob (~ed@2a00:d880:6:320:82fa:b33e:3d20:4763) |
| 21:05.39 | [TK]D-Fender | SHOW ME THE CALLS |
| 21:05.41 | [TK]D-Fender | ^^^^^^^^^ |
| 21:05.45 | [TK]D-Fender | and the LIVE STATUS DUMPS |
| 21:05.51 | [TK]D-Fender | I want proof about what's happening. |
| 21:05.55 | [TK]D-Fender | and re-PB your configs |
| 21:06.28 | ModFather | http://pastebin.ca/3733531 ( queues.conf ) |
| 21:06.37 | WIMPy | Make sure your call counters are enabled for those peers. |
| 21:06.46 | igcewieling | ModFather: try the CLI |
| 21:09.26 | ModFather | extensions.conf http://pastebin.ca/3733532 |
| 21:10.42 | WIMPy | Check 'sip show inuse' while on a call. If you don't see the peers are busy there, check your sip.conf. |
| 21:11.01 | igcewieling | ModFather: try the CLI |
| 21:11.14 | *** join/#asterisk v1r0 (~viro@unaffiliated/viro) |
| 21:13.17 | ModFather | sorry this: http://pastebin.ca/3733535 |
| 21:14.34 | igcewieling | Wow, maybe he is genetically incapable of providing console log info? |
| 21:15.24 | igcewieling | You really should stop wasting everyone's time. |
| 21:15.26 | ModFather | http://pastebin.ca/3733536 |
| 21:15.44 | ModFather | i was just make sure i have PB the configurations |
| 21:15.53 | ModFather | and then i just made a call |
| 21:16.06 | igcewieling | [TK]D-Fender: he finally posted CLI output |
| 21:17.32 | [TK]D-Fender | <WIMPy> Make sure your call counters are enabled for those peers. |
| 21:17.39 | [TK]D-Fender | And in between I asked for..... |
| 21:17.45 | [TK]D-Fender | <[TK]D-Fender> and the LIVE STATUS DUMPS |
| 21:17.57 | [TK]D-Fender | queue show |
| 21:17.58 | [TK]D-Fender | ^ |
| 21:18.09 | [TK]D-Fender | callcounter=yes <- sip.conf entries |
| 21:18.23 | ModFather | i do not have callcounter=yes on my sip.conf |
| 21:18.33 | ModFather | really sorry for wasting your time guys, still learning.. |
| 21:26.29 | ModFather | http://pastebin.ca/3733538 LIVE STATUS DUMPS ( queue show ) |
| 21:27.40 | ModFather | after i answer from 101 it was still saying ( Not in use ) |
| 21:28.11 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:28.14 | [TK]D-Fender | I expect to see that being called INLINE |
| 21:28.17 | [TK]D-Fender | You are missing the point. |
| 21:28.22 | [TK]D-Fender | I want to see it ALL happening |
| 21:28.38 | Ecco | Hey |
| 21:28.44 | ModFather | how i can output all that you want to see [TK]D-Fender ? |
| 21:28.47 | Ecco | With the same callfile |
| 21:28.57 | Ecco | I'm sometimes getting an "Forbidden" response from Asterisk |
| 21:28.58 | [TK]D-Fender | .... |
| 21:29.01 | Ecco | How can I figure out why? |
| 21:29.28 | [TK]D-Fender | Go find out what the other side is asking... go look at what your side is set to accept. |
| 21:29.30 | [TK]D-Fender | rectify that |
| 21:29.56 | Ecco | well, the exact same callfile works like 50% of the time |
| 21:30.00 | Ecco | same recipient, same everything |
| 21:30.36 | [TK]D-Fender | go look at the call |
| 21:30.47 | [TK]D-Fender | callfile doesn't magically mean anything to a SIP CALL |
| 21:31.01 | Ecco | how? where? |
| 21:31.24 | [TK]D-Fender | CLI + SIP debug clearly |
| 21:33.12 | ModFather | [TK]D-Fender after i added callcounter=yes to [101] and [102] sip.conf worked |
| 21:37.07 | ModFather | [TK]D-Fender thank you |
| 21:37.10 | ModFather | igcewieling thank you |
| 21:37.16 | ModFather | WIMPy thanks man |
| 21:53.35 | *** join/#asterisk n3ob (~ed@2a00:d880:6:320:82fa:b33e:3d20:4763) |
| 22:00.15 | dan_j | When using realtime, why would astcanary often not run? I regularly get "Asterisk PBX is running but astcanary is not although it should" from pacemaker |
| 22:00.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 22:00.44 | dan_j | Only on one of the servers in the pair. The other one runs properly every time. |
| 22:01.32 | dan_j | And is astcanary/realtime preferred/recommended or just optional? |
| 22:06.23 | dan_j | Alternatively, where can I find log files for astcanary? |
| 22:16.33 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 22:20.44 | wdoekes | dan_j: https://issues.asterisk.org/jira/browse/ASTERISK-26352 <-- astcanary dies after core restart |
| 22:22.28 | wdoekes | as for using it, I would say optional, I've never once needed its services. but as always, YMMV |
| 22:26.27 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 22:26.58 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 22:28.10 | *** join/#asterisk CheBuzz (~CheBuzz@unaffiliated/chebuzz) |
| 22:31.48 | kfife | there's an standard channel variable for the extension |
| 22:31.52 | kfife | Oops |
| 22:36.20 | dan_j | wdoekes: I'm using v13 so I assume this bug has been fixed in that. |
| 22:36.55 | dan_j | I'll disable it and see if i can live without it. still not quite sure why it's required if asterisk runs fine without rt |
| 22:39.50 | *** join/#asterisk n3ob (~ed@2a00:d880:6:320:82fa:b33e:3d20:4763) |
| 22:52.22 | kfife | Is there a standard channel variable, or function that I can use to determine the previous extension? For example, in an IVR, if the i extension is called, I want to return to a sane place in the previous extension. Ideas? |
| 23:01.14 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 23:02.59 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 23:06.35 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 23:10.25 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 23:14.49 | [TK]D-Fender | kfife, It's your context. There is no special "previous place". You jump where you want to |
| 23:16.20 | kfife | So in the IVR application, I would need to manually track ${THISCONTEXT} ${LASTVALID} etc |
| 23:16.30 | kfife | oops |
| 23:16.34 | kfife | not thiscontext |
| 23:16.37 | kfife | that's standard. |
| 23:16.47 | kfife | meant to say LASTCONTEXT |
| 23:16.55 | kfife | LASTVALID etc |
| 23:16.59 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 23:18.31 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
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| 23:51.04 | [TK]D-Fender | It's your IVR... how do you not know where to jump to? |
| 23:51.35 | [TK]D-Fender | If I give you a prompt saying to do 1 of 5 things.... and hit an invalid option.... it's MY menu. I know where I want to jump to. |
| 23:51.38 | [TK]D-Fender | Thre is no"last". |
| 23:51.47 | [TK]D-Fender | Wjhy would there be 5 places I could have come from? |
| 23:51.51 | [TK]D-Fender | that makes no sense |
| 23:52.09 | [TK]D-Fender | "last isn't a thing with a point to it at all. |
| 23:52.32 | [TK]D-Fender | YOU put the WaitExten or Background that allowed input in the first palce |
| 23:52.34 | [TK]D-Fender | place |
| 23:52.45 | [TK]D-Fender | So YOU should already know where you want to jump to |
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