| 00:18.56 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:18.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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 |
| 00:30.46 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 00:37.51 | Samot | All but ones of those costs money |
| 00:38.08 | Samot | All but one of those will likely license you by channels. |
| 00:39.12 | Samot | vader-: Also, what are they looking for? What features? What extras? Etc, etc.... |
| 00:39.30 | Samot | What are the comparison points you need to check? |
| 00:41.45 | vader- | They sent me a sheet to try and fill out. They want like live monitoring and recorsing |
| 00:41.48 | vader- | Recording |
| 00:42.20 | vader- | Their biggest thing they want I guess is non technical people to be able to setup devices, users, queues, etc. |
| 00:44.26 | Samot | Live Monitoring? |
| 00:44.28 | Samot | Of? |
| 00:45.00 | vader- | Agents in the queue |
| 00:45.12 | vader- | They want to listen to the calls |
| 00:45.14 | vader- | Live |
| 00:45.24 | Samot | OK so with FreePBX you would need add-ons. |
| 00:45.35 | Samot | As for the others you would need to look to see what they supported. |
| 00:46.10 | Samot | And you need an endpoint manager |
| 00:46.10 | vader- | Other then the systems I mentioned is there any you recommend? |
| 00:46.25 | Samot | I don't use Avaya, Mitel or Cisco. |
| 00:46.51 | Samot | Or really 3CX because they cost money just to fire them up. |
| 00:47.48 | vader- | What's involved with configuring free PBX for high availability |
| 00:48.16 | Samot | OK so that's no longer supported |
| 00:48.21 | Samot | Because it barely worked before. |
| 00:48.28 | Samot | But HA is a pretty big requirement |
| 00:48.33 | Samot | Should have been on that list. |
| 00:48.38 | Samot | Like top of it. |
| 00:48.40 | vader- | <PROTECTED> |
| 00:48.57 | Samot | I meant the one you just gave to me. |
| 00:49.08 | vader- | Oh Hehe |
| 00:49.14 | Samot | Because the other points are not worth looking at if the HA is a requirement. |
| 00:49.27 | Samot | Now you need to find a vendor that supports that type of setup |
| 00:49.33 | vader- | I guess free PBX could do passive HA |
| 00:49.36 | Samot | Which will require _2_ licenses for everything. |
| 00:49.39 | vader- | Just have a cold standby |
| 00:49.53 | vader- | With the config and licenses loaded |
| 00:49.56 | Samot | Even Passive HA requires some work on it. |
| 00:50.08 | Samot | Doesn't matter which vendor you go with |
| 00:50.20 | Samot | If you have an HA/Warm Spare setup, you need two of everything. |
| 00:51.24 | vader- | Ya |
| 00:52.20 | Samot | So right now jumping from FreePBX to those other vendors (even with some of the commercial modules) means you're jumping about $1500 dollars. |
| 00:52.32 | Samot | 3CX is going to be about $1500 a year |
| 00:52.45 | Samot | Cisco, Avaya, Mitel..probably more. |
| 00:53.01 | Samot | Well $1500 up front and then some serious yearly fees. |
| 00:53.26 | vader- | I was just ball parking this might run them $1000/channel |
| 00:53.30 | Samot | Is there an upper budget for this? |
| 00:53.37 | vader- | So like $35k |
| 00:53.41 | Samot | No, not at all |
| 00:54.00 | Samot | $35K is what LEGACY goes for. |
| 00:54.05 | vader- | It was a high ballpark |
| 00:54.10 | *** join/#asterisk cleronx (cleron@s206-116-6-158.bc.hsia.telus.net) |
| 00:54.42 | Samot | You're going to have to get quotes. |
| 00:54.51 | Samot | Most of those vendors don't just slap their pricing on their website. |
| 00:55.16 | vader- | Ya |
| 00:55.39 | vader- | Figured new handsets will run like $100-200 ea |
| 00:55.57 | vader- | So there is 10-20% of the $1000/per agent cost |
| 00:56.02 | Samot | Depends on the handsets needed but that's a fair ballpark |
| 00:56.09 | Samot | Well |
| 00:56.20 | Samot | It's like $4K or so for the upper licenses for 3CX |
| 00:56.27 | Samot | I think those others would be about the same park |
| 00:56.39 | Samot | It's not going to be $35K |
| 00:57.11 | vader- | That price includes consultants for installation, configuration and training |
| 00:57.12 | Samot | Not when the Mitel legacy PBX systems run $25K+ |
| 00:57.14 | vader- | As well |
| 00:57.32 | Samot | Mitel's _legacy_ PBX's go for that price. |
| 00:57.41 | Samot | There is no way their IP PBXs are going to be in that range. |
| 00:57.46 | Samot | Makes absolutely no sense. |
| 00:59.45 | Samot | Have you gone to any of these vendors are looked to see if they have pricing for their PBX systems or licenses for the software? |
| 00:59.50 | Samot | s/are/and/ |
| 01:02.23 | cleronx | hey im trying to get my astrisk based appointment reminder system to dial out a call to my FreePBX but i keep getting error message anyone know what i did wrong? |
| 01:02.24 | cleronx | <PROTECTED> |
| 01:02.25 | cleronx | <PROTECTED> |
| 01:04.10 | Samot | Is ib_dial call file using ib_dial for the Channel variable or for the context variable? |
| 01:04.48 | cleronx | not quite sure, how would i check that? |
| 01:04.55 | Samot | You wrote it? |
| 01:05.00 | Samot | Correct? |
| 01:05.42 | cleronx | no it was configured like this when i bought the software. i havent used it for a while. now that i got it up and running again it wont dial out |
| 01:06.12 | Samot | What is the software? |
| 01:06.17 | vader- | I am just starting the process |
| 01:06.19 | cleronx | ICT Broadcast |
| 01:06.42 | vader- | Figured I would just pose the question here first and see if anyone has done anything similar lately. |
| 01:07.19 | Samot | vader-: Everyone needs are different. |
| 01:08.11 | vader- | True but they might have some.overlap |
| 01:08.28 | cleronx | What are you trying to do Vader/ |
| 01:08.30 | cleronx | ? |
| 01:10.33 | cleronx | anyone an esterisk exper here? |
| 01:10.40 | cleronx | asterisk expert? |
| 01:11.18 | Samot | cleronx: What versions of ASterisk does this software support? |
| 01:12.20 | Samot | vader-: All those vendors have UC system products as PBX/voice system offerings. |
| 01:12.38 | cleronx | right now its running: Asterisk 1.8.18.0 built by mockbuild @ buildvm-09.phx2.fedoraproject.org on a x86_64 running Linux on 2012-11-08 |
| 01:13.07 | Samot | You're going to need to look at their offerings to figure out what ones cross over, which ones have certain features or doesn't have them and then decide which one closely meets the requirements you have. |
| 01:13.22 | Samot | Asterisk 1.8 is not supported. |
| 01:13.45 | cleronx | it was working before fine. just havent used it for about a year. not sure if i changed something on my pbx or what |
| 01:14.01 | cleronx | its not supported where? |
| 01:14.28 | Samot | Asterisk 1.8 is EOL |
| 01:14.34 | Samot | Has been for a few years now. |
| 01:15.08 | cleronx | yah thats ok. It was working fine before. so the software installed is till functional |
| 01:15.12 | cleronx | still |
| 01:15.17 | Samot | But it doesn't work now. |
| 01:16.00 | cleronx | are you saying that the astrisk version blocks it self from working after a certin time? |
| 01:16.06 | Samot | No. |
| 01:16.23 | Samot | What I am saying is you're asking for help on a version of Asterisk that no longer has real support. |
| 01:16.27 | Samot | By Digium... |
| 01:16.53 | Samot | Things have seriously changed since 1.8 |
| 01:17.05 | Samot | A "current day" Asterisk expert might not have 1.8 experience. |
| 01:17.08 | cleronx | but there are still experts that know how to use it and remember the comands |
| 01:17.16 | Samot | OK |
| 01:17.19 | Samot | So I asked you a question |
| 01:17.24 | Samot | You didn't know the answer. |
| 01:17.38 | Samot | Since this is some third part software that is generating this you need to find out how the third party is doing it. |
| 01:17.49 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 01:18.28 | Samot | You've just shown a failed context, where a call goes when it fails to Dial() |
| 01:18.44 | cleronx | well they have there own gui but it runs on asterisk. Its asterisk that has to be modified to make sure it connects to my pbx properly |
| 01:18.45 | Samot | For whatever reason, you should show the actual call being made before it fails. |
| 01:19.38 | cleronx | i can do that, just dont know how to find that information. Im not an expert asterisk. is there a command i can type to vew this information? |
| 01:19.50 | Samot | This isn't Asteirsk |
| 01:19.58 | Samot | This is ICT Broadcast |
| 01:20.24 | *** join/#asterisk zigggggy (ffssd@unaffiliated/zigggggy) |
| 01:20.43 | Samot | That is the software generating the call files and the originate command to make the call out Asterisk. |
| 01:20.45 | Samot | It's failing. |
| 01:20.48 | cleronx | i wasint sure if it was ict broadcast that was not sending the call to asterisk or asterisk not sending to my pbx |
| 01:21.02 | Samot | I don't know |
| 01:21.07 | Samot | You haven't shown a whole call. |
| 01:21.25 | cleronx | how can i shouw the whole call. i would like to |
| 01:21.39 | Samot | asterisk -rvvvvvvvvvvv |
| 01:21.41 | Samot | Make call |
| 01:21.52 | Samot | Post output from the cli to pastebin |
| 01:22.00 | cleronx | ok |
| 01:22.10 | cleronx | what is pastebin? |
| 01:23.23 | Samot | ~pb |
| 01:23.24 | infobot | pastebin is probably a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:24.53 | cleronx | http://pastebin.ca/4033999 |
| 01:25.33 | Samot | -- Executing [s@macro-call_failed:4] ExecIf("OutgoingSpoolFailed", "1?Set(response=hangup)") in new stack |
| 01:25.43 | Samot | So your other PBX hung up the call. |
| 01:25.56 | Samot | Now you need to look at that PBX as to why it hung up |
| 01:26.55 | cleronx | but when i do asterisk -rvv on the pbx. it does not show anything when i place the call. and on ICT Broadcast it starts going threw the numbers really fast as if it wasint connecting at all |
| 01:27.25 | cleronx | its almost like Freepbx is not getting the call at all |
| 01:27.55 | cleronx | also on ICT Broadcast it sais the trunk is unrechable |
| 01:28.07 | cleronx | but on the freepbx it shows that it is registered |
| 01:28.36 | Samot | The the Asterisk server running ICT show the trunk in an OK status? |
| 01:28.58 | Samot | Just because the one says it is registered doesn't mean the other thinks it's still alive where the registrations says it would be. |
| 01:29.32 | cleronx | yah on ICt Broadcast it seas unrechable |
| 01:30.52 | Samot | So now you need to figure out why |
| 01:31.08 | cleronx | yap |
| 01:31.16 | Samot | that generally means the ICT server isn't getting responses back from the other PBX. |
| 01:31.24 | cleronx | thats what i was hoping to get some help here with |
| 01:31.34 | Samot | No it wasn't. |
| 01:31.38 | Samot | You just figured it out. |
| 01:31.44 | cleronx | i did ? |
| 01:31.45 | Samot | http://xyproblem.info |
| 01:31.54 | Samot | You had no clue the trunk was unreachable |
| 01:32.05 | cleronx | yes i did |
| 01:32.10 | Samot | You never once said that |
| 01:32.18 | Samot | You said you wanted to know why calls were failing. |
| 01:32.24 | Samot | Well they are failing because the trunk is unreachable |
| 01:32.39 | cleronx | ohh ok, sorry i explained it wrong. |
| 01:32.44 | cleronx | hmm |
| 01:32.47 | Samot | http://xyproblem.info |
| 01:35.03 | Samot | Is this FreePBX system on the same network as the ICT system or is this going over the Internet? |
| 01:35.11 | cleronx | yah read the site... lol. thats becouse most people are not experts and dont really know how to explain there problem and the details that an expert needs to solve the problem |
| 01:35.19 | cleronx | nop |
| 01:35.20 | cleronx | nope |
| 01:35.25 | Samot | To which? |
| 01:35.40 | cleronx | freepbx is on a hosted server and ict is in my office |
| 01:35.45 | Samot | OK |
| 01:35.54 | Samot | Is there a firewall where the FreePBX server is? |
| 01:35.59 | Samot | Or is it behind NAT? |
| 01:36.30 | cleronx | hmm.. i have a static ip i dont think there is a firewall. but ict is on a nat router |
| 01:36.49 | Samot | OK |
| 01:37.04 | Samot | So either the NAT for the network the ICT server is on is messed up |
| 01:37.19 | Samot | Meaning the replies from the FreePBX are not making it back to the ICT server. |
| 01:37.29 | Samot | Or the requests from the ICT are not making it to the FreePBX |
| 01:37.45 | Samot | So this is a networking issue. I'm going to go with the NAT is probably messed up. |
| 01:37.46 | cleronx | yah i belive the requests from the ICT are not making it to the FreePBX |
| 01:38.05 | Samot | OK still a networking problem, just where the FreePBX server is. |
| 01:38.06 | cleronx | becous nothing is showing on my pbx |
| 01:38.12 | cleronx | when a call is laced |
| 01:38.18 | cleronx | placed |
| 01:39.26 | cleronx | well i have been in the process of moving my freepbx in to the office so i do have a copy of it in virtual box. i can try running it on the same network and see if it still causes the problem |
| 01:44.19 | *** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net) |
| 01:44.32 | cleronx | ohh god... i started a copy of the vm and its working if i set it up as a trunk |
| 01:44.38 | cleronx | i guess nat is the problem |
| 01:48.30 | Samot | Or something else. |
| 01:48.37 | Samot | But it's 100% networking at this point |
| 02:14.13 | [J]oules | hi, after running yum update and installing asterisk-13.19.1 endpoints using ipv6 no audio |
| 02:14.55 | [TK]D-Fender | This is #asterisk |
| 02:14.58 | [J]oules | rtp debug does show its sending to correct ipv6 address |
| 02:15.11 | [TK]D-Fender | Who don't particularly care about unofficial repos |
| 02:15.32 | [TK]D-Fender | and non-current releases |
| 02:16.03 | [TK]D-Fender | #wrongtree |
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| 02:32.52 | Samot | #shotgunhelp really since he made the same statements in both channels. |
| 02:33.11 | Samot | #poorfollowup can go with that since there is zero followup. |
| 02:40.38 | drmessano | I may have a good consultant who can help [J]oules |
| 02:40.53 | drmessano | He's chinese but he speaks pretty decent english |
| 02:41.00 | drmessano | His name is Sum Ting Wong |
| 02:41.07 | drmessano | I'll send his info |
| 02:41.48 | [J]oules | drmessano: Ahh,. i thought it would be something like yun fing messano |
| 02:42.53 | drmessano | That would imply a member of my family would ever help you |
| 02:43.14 | [J]oules | no thanks |
| 02:43.18 | drmessano | You must be Buffala Crazy |
| 02:43.24 | [J]oules | ignore time |
| 02:43.33 | drmessano | Please do |
| 02:43.34 | Samot | HAHA |
| 02:43.53 | [J]oules | ignore time |
| 02:43.57 | [J]oules | *!drmessano@pdpc/supporter/active/drmessano |
| 02:44.00 | drmessano | Please do |
| 02:44.05 | Samot | Dude, were you just informing us that you had a problem after updating Asterisk or were you looking for someone to help with the issue? |
| 02:44.25 | drmessano | He can't fucking ever do anything correctly |
| 02:44.29 | drmessano | Even ignoring someone on IRC |
| 02:44.43 | drmessano | Please let me give you my money for services |
| 02:45.08 | drmessano | I trust you with my 911 calls, really |
| 02:45.24 | drmessano | #NowDatsScaryAF |
| 02:46.00 | Samot | Now, now. It's cool. Only "his" box is running PJSIP over IPv6 so none of his customers would have this issue. |
| 02:46.13 | Samot | Or at least that's the story from before. |
| 02:46.23 | drmessano | Well |
| 02:46.34 | drmessano | To be fair, you literally can't believe anything he says, ever |
| 02:46.37 | Samot | Sorry, PJSP/TLS/IPv6 |
| 02:47.42 | drmessano | Well, Samot |
| 02:47.58 | drmessano | PJSIP+TLS+IPV6 is the Buffala VoIP Stack |
| 02:48.01 | drmessano | PJSIP+TLS+IPV6 is the Buffala VoIP Stack (TM) |
| 02:49.12 | drmessano | If you like, you can call his sales department where he pretends to be 3 different people, and sells you on the virtues of his NATProof Buffala VoIP Stack |
| 02:49.46 | [J]oules | Samot: no, yes a problem, but the issue happened right after upgrading asterisk to 13.19.1 |
| 02:49.55 | *** join/#asterisk iq (~iq@47.185.85.95) |
| 02:50.07 | drmessano | yum updata buffala-voip-stack |
| 02:50.14 | Samot | AND? |
| 02:50.24 | Samot | Do you have a call to show that has no audio? |
| 02:50.50 | [J]oules | the issue is no audio on endpoint, go to endpoint change network to 'prefer ipv4, end of audio issue |
| 02:51.02 | Samot | Sigh |
| 02:51.09 | Samot | So using a completely DIFFERENT thing works |
| 02:51.18 | [TK]D-Fender | That isn't debugging |
| 02:51.31 | Samot | That's like saying "My VCR doesn't blink anymore, I put electrical tape over it" |
| 02:51.39 | [J]oules | someone is testing it out now that is well versed in this |
| 02:51.50 | drmessano | He's got Lanny helping him now |
| 02:51.53 | [J]oules | i dont have a vcr |
| 02:52.00 | [J]oules | that is for rich people like you |
| 02:52.25 | drmessano | Rich? He's the one with IPv6 phones |
| 02:52.32 | drmessano | Mine don't even do IPv6 |
| 02:52.40 | drmessano | :((((( |
| 02:54.17 | drmessano | Besides, this is off-topic for this channel anyway. He should be asking in #asterisk-ipv6-tls-pjsip |
| 02:54.23 | drmessano | It's a dedicated channel |
| 02:56.41 | Samot | I understand. |
| 02:56.48 | Samot | You can't afford a VCR with all those MS licenses. |
| 02:57.06 | drmessano | HA |
| 02:57.12 | drmessano | ^ He's right you know |
| 02:57.55 | Samot | When you want to compare how "rich" the companies are, you're going to win. |
| 02:58.14 | Samot | You have more assets in play than I do. |
| 02:58.51 | drmessano | He said "ass".. huh huh huh huh |
| 03:00.31 | Samot | So if I'm following this properly. You came in here and asked for help in TWO channels. When you get offered help and are told to provide a debug then answer is "I got someone experienced working on it". |
| 03:00.45 | Samot | So why did you need to ask for help in two channels if you have someone experienced working on it? |
| 03:02.31 | [TK]D-Fender | So he can play both side against each other |
| 03:02.39 | [TK]D-Fender | on the assumption that other side even exists |
| 03:03.28 | Samot | I thought maybe it was like a race. Like the other person bet him he couldn't figure out the issue before them. |
| 03:04.20 | [TK]D-Fender | #mantiteo |
| 04:10.42 | drmessano | I wonder how many personalities he has |
| 04:13.43 | Samot | Lol |
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| 13:39.35 | [J]oules | again confirmed no audio when using ipv6 from at least bria in smartphone since upgrade of asterisk to 13.19.1 |
| 13:45.04 | Samot | And? |
| 13:47.11 | Samot | What happened with the testing from last night |
| 13:47.32 | Samot | Where are any debugs showing a call with audio issues (as requested last night)? |
| 13:47.48 | [TK]D-Fender | Stop talking like that version is cursed without providing backup. |
| 13:48.34 | [TK]D-Fender | And remember that even if there was a problem with it, it isn't the current release and you'd have to upgrade and reproduce and provide all of that debug instead. |
| 13:49.03 | Samot | Well that's going to be the rub. |
| 13:49.17 | Samot | FreePBX, man. 13.19.1 is the "latest" Asterisk version in the repos. |
| 13:49.24 | Samot | For 13 |
| 13:49.59 | [TK]D-Fender | [TK]D-Fender |
| 13:50.00 | [TK]D-Fender | Who don't particularly care about unofficial repos |
| 13:50.00 | [TK]D-Fender | 10:15 pm |
| 13:50.00 | [TK]D-Fender | and non-current releases |
| 13:50.00 | [TK]D-Fender | 10:16 pm |
| 13:50.00 | [TK]D-Fender | #wrongtree |
| 13:50.38 | Samot | I get that. I'm just informing you that your suggestion of getting on 13.21 isn't going to happen. |
| 13:50.44 | *** join/#asterisk Sepultura (~Sepultura@unaffiliated/sepultura) |
| 13:51.01 | [TK]D-Fender | And I stated that this channel doesn't care unless he does so. #winwin |
| 13:51.29 | Samot | And when he asks in #freepbx and it is the current? |
| 13:52.59 | [TK]D-Fender | He can go get all the debug he's been asked for repeatedly, go check the tracker to see if there was a reported issue if the (proper) debug doesn't explain anything. and then this channel will be quiet :) |
| 13:53.06 | [TK]D-Fender | #halfwaythere |
| 13:53.39 | [TK]D-Fender | thinks 41,700,000ms is a hard tail to EC..... |
| 13:54.43 | Sepultura | file, Am I right if a Company has a MSDN Subscription it gets Windows OS'es for free? |
| 13:54.57 | file | blinks |
| 13:56.15 | Samot | rofl |
| 13:56.20 | Samot | That was total left field. |
| 13:57.45 | file | quite |
| 13:58.16 | Samot | Sepultura: Are you planning to do development on a MS OS? |
| 13:58.42 | Sepultura | Samot the most companies are using Windows |
| 13:58.48 | Samot | Yes. |
| 13:58.57 | Samot | MSDN = Microsoft Developers Network |
| 13:59.00 | [TK]D-Fender | Why are you asking HERE? |
| 13:59.07 | Samot | It's for reviewing and developing MS OS software. |
| 13:59.15 | Samot | It's not meant for production use. |
| 13:59.55 | Sepultura | because file is a developer |
| 14:00.02 | Samot | So am I. |
| 14:00.13 | Samot | Just not a closes source system developer. |
| 14:00.19 | Samot | closed |
| 14:00.28 | Sepultura | Samot you are allowed to code OSS for WIndows |
| 14:00.33 | Samot | OK |
| 14:00.52 | Samot | MSDN is for developers. It gets you a _single_ copy of the lastest software/OSes for MS. |
| 14:00.58 | Samot | That cannot be used in production. |
| 14:02.38 | Sepultura | Samot then I wonder why ebay still allows sellers to sell MSDN Licenses o.O |
| 14:05.16 | Samot | I don't know. |
| 14:05.30 | Samot | But you should look at Microsoft's site for answers on the MSDN |
| 14:08.33 | Sepultura | Samot you are from US? |
| 14:09.03 | Samot | Yes |
| 14:09.31 | Sepultura | so you are MSc of Computer Science? |
| 14:09.36 | Sepultura | or Selfmade Coder? |
| 14:10.54 | Samot | The latter. |
| 14:11.46 | Sepultura | you are working as Coder too? |
| 14:12.10 | Samot | Technically. Since I write code for my own platforms. |
| 14:12.18 | Samot | And all the stuff that I need to do. |
| 14:12.39 | *** join/#asterisk SebastienThiry (~Thunderbi@91.180.9.33) |
| 14:12.42 | Sepultura | you own an company or are you an employee? |
| 14:12.50 | Samot | What is this? |
| 14:13.06 | Samot | I've been very open, I run my own shop. |
| 14:13.08 | Sepultura | employee? |
| 14:13.19 | Sepultura | do you have a boss? |
| 14:13.21 | Samot | Does it matter? |
| 14:13.25 | *** join/#asterisk imcdona (~imcdona@2607:f0d8:20:1001:e4d6:c683:1a7d:6045) |
| 14:13.26 | Samot | Yes. My wife. |
| 14:13.28 | Sepultura | or are you your own boss? |
| 14:13.36 | Samot | But that has nothing to do with business. |
| 14:14.01 | Samot | 10:13:04 AM <Samot> I've been very open, I run my own shop. |
| 14:14.06 | Sepultura | I heard it is hard to get a job with just High School |
| 14:14.09 | Sepultura | diploma |
| 14:14.16 | Samot | Only if you're an idiot. |
| 14:14.16 | Sepultura | I am from Germany |
| 14:14.29 | Samot | It's also hard to get a job with a college/university degree. |
| 14:14.41 | Sepultura | so you can actually get a decent job with High School Diploma? |
| 14:14.46 | Samot | Yes. |
| 14:16.40 | Samot | In a lot of cases jobs will say "A degree in X OR Y amount of years in experience" |
| 14:18.29 | Sepultura | I need to go. In an Hour I will be back |
| 14:18.31 | Sepultura | cya |
| 14:22.14 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 14:22.43 | *** join/#asterisk lilDangerz (~lildanger@fixed-187-189-68-170.totalplay.net) |
| 14:37.21 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-piyuwfgztsqolenr) |
| 14:37.21 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:39.21 | igcewieling | I'm considering moving Asterisk log info from a text file (/var/log/asterisk/full, etc) into a database. That should make the log info easier to search. Has anyone here done something similar and if so, how did it work out? |
| 14:50.19 | rrittgarn | I hadn't ever considered it, but if i was going to, I would probably toss it into ELK instead of a straight DB |
| 14:50.47 | rrittgarn | at the volume I create full logs, I'd be afraid that i'd overload a DB if i were doing it live |
| 14:55.38 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-mnrzyxptmqvmpkwu) |
| 14:55.38 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:06.46 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 15:30.27 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 15:41.06 | DanQuinney | igcewieling: using graylog or rolling your own? |
| 15:41.25 | DanQuinney | we use the former and it works well |
| 15:42.22 | igcewieling | I've already rolled my own which uses the flat files. |
| 15:43.10 | igcewieling | Most of the time it is easy to parse messages from the logs -- except for the log entries which use more than one line like Dumpchan. |
| 15:45.23 | igcewieling | DanQuinney: my script takes the dialplan and the log entries and then matches the two so in addition to seeing GotoIf(0?8:21) in the log, I see ["${GLOBAL(${breaker})}" == "tripped"]?8:21 which shows me the log line before it was evaluated. |
| 15:45.44 | igcewieling | VERY handy for debugging. |
| 15:48.28 | igcewieling | sort of like this: https://i.imgur.com/zLmbSF0.png |
| 15:54.35 | DanQuinney | yes, I could see that being handy |
| 16:29.58 | igcewieling | rrittgarn: good point. |
| 16:47.40 | *** join/#asterisk Kaian (~kaian@212.81.221.228) |
| 17:39.19 | [J]oules | confirmed polycom set to 'prefer ipv6' has no audio on calls. Rebooting it now with network setting to ipv4 only |
| 17:42.00 | [TK]D-Fender | Still wasting our time.... |
| 17:42.16 | [TK]D-Fender | You're not showing actual debug from the current release. |
| 17:44.10 | [J]oules | i dont debug what I dont hear :) |
| 17:44.34 | [J]oules | rebooted polycom with network set to ipv4 only, redial same TN, audio works |
| 17:44.38 | [TK]D-Fender | And what we can't see we don't care about. |
| 17:44.58 | [J]oules | you're in the telco business, you don't like sounds? |
| 17:45.05 | [TK]D-Fender | So for an anecdotal story ... that's nice :) |
| 17:45.14 | [J]oules | yah i know, we are working on it |
| 17:45.21 | [J]oules | sip debug |
| 17:45.28 | drmessano | So why tweet in here about it |
| 17:45.33 | drmessano | If "we" are working on it |
| 17:45.38 | drmessano | We don't need to hear all about your day |
| 17:46.12 | drmessano | "I just pooped" "Rebooting my phone" "Checked Reddit, again" |
| 17:46.52 | [TK]D-Fender | "Still looking (not really) at something I'll claim is a legit issue unrelated to my inability to manage my systems..." |
| 17:47.22 | drmessano | What he basically just confirmed is that he has a non-working IPv6 configuration |
| 17:47.41 | drmessano | Still not sure how that isn't a problem for #networking |
| 17:48.20 | [TK]D-Fender | Could be |
| 17:48.50 | [TK]D-Fender | Could be a lot of things, none of which are worth wasting time guessing about of course |
| 17:50.37 | [TK]D-Fender | wonders how many hours/days/half-assed-attempts it'll take to actually get something appropriate and worthwhile.... |
| 17:50.46 | Samot | It's like "Keystone Tech" |
| 17:52.20 | drmessano | Ok, when we start making guesses and see who is later correct? |
| 17:53.51 | [TK]D-Fender | Or where you dress up in vintage police uniforms and perform in a silent film with copious amounts of physical comedy? |
| 17:54.18 | drmessano | We could do that too |
| 17:55.19 | drmessano | Im guessing he goofed his IPv6 config prior to updating Asterisk. Since he likes to reboot 2 or 3 times for everything, network changes took effect on reboot |
| 17:55.41 | drmessano | I bet he can't ping6 the host |
| 18:01.26 | [TK]D-Fender | When in doubt burn the evidence |
| 18:04.50 | *** join/#asterisk SebastienThiry (~Thunderbi@217.82-65-87.adsl-dyn.isp.belgacom.be) |
| 18:11.06 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 18:16.38 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 18:20.29 | [J]oules | https://www.irccloud.com/pastebin/LgLDZF4o/ |
| 18:22.46 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 18:24.11 | [J]oules | https://www.irccloud.com/pastebin/UqozCcZf/ |
| 18:24.51 | [J]oules | so where do you see in the p/b the audio problem?? |
| 18:29.28 | Samot | Well all I see here is one side of the call. |
| 18:29.36 | [J]oules | the call above in p/b was recorded, playback and there is audio |
| 18:29.36 | Samot | Between the PBX and your "billing" server. |
| 18:29.45 | Samot | I don't see anything between the endpoint and the PBX. |
| 18:30.09 | [J]oules | Samot: pasted what was in cli |
| 18:30.19 | [J]oules | yeah changed ip's etc. but thats it |
| 18:30.24 | Samot | Then you didn't go back far enough |
| 18:30.33 | Samot | You don't have the INVITE from the endpoint/phone to the PBX |
| 18:30.39 | Samot | You have the call between your two systems. |
| 18:30.44 | [J]oules | untrue, have cli open and then hit redial on phone |
| 18:30.50 | Samot | OK |
| 18:31.00 | *** join/#asterisk K0HAX (~michael@28.139.154.104.bc.googleusercontent.com) |
| 18:31.03 | [TK]D-Fender | https://www.irccloud.com/pastebin/UqozCcZf/ <- don't see initial invite |
| 18:31.04 | [J]oules | core set verbose 5 |
| 18:31.05 | Samot | Then all I see is a call between two FreePBX boxes. |
| 18:31.21 | [TK]D-Fender | https://www.irccloud.com/pastebin/LgLDZF4o/ <- deleted |
| 18:31.25 | Samot | I don't see anything from the endpoint making the call to the FreePBX |
| 18:31.29 | [J]oules | so what then do you want to see? |
| 18:31.36 | [J]oules | ok try it again |
| 18:31.37 | [TK]D-Fender | THE ENTIRE CALL |
| 18:31.44 | Samot | Well since the endpoint is the one with no audio.... |
| 18:32.07 | [J]oules | no [TK]D-Fender just that p/b, the one above is same but ip's changed to protect the guilty |
| 18:32.22 | *** join/#asterisk mike543 (404ac60a@gateway/web/freenode/ip.64.74.198.10) |
| 18:32.23 | [TK]D-Fender | that non-deleted PB has no INITIAL debug |
| 18:32.49 | [J]oules | [TK]D-Fender: you want to tv in and watch for yourself? |
| 18:33.22 | Samot | So the person that was testing found nothing? |
| 18:33.26 | [TK]D-Fender | I want you not to be incompetent and actually post something properly |
| 18:33.31 | Samot | Couldn't turn up anything as to why this is happening? |
| 18:33.48 | [J]oules | neither can i find what it is |
| 18:33.50 | [TK]D-Fender | Samot, I would assume not since we're all still here at it |
| 18:34.01 | *** join/#asterisk cresl1n (uid299068@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 18:34.01 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:34.16 | [J]oules | the audio was most definietly working because my pbx recorded it and i played it back |
| 18:34.19 | [TK]D-Fender | Get a complete set of debug |
| 18:34.28 | [J]oules | its just not getting from pbx to endpoint |
| 18:35.02 | Samot | Wait. Are you on the right box? |
| 18:35.18 | [J]oules | yeahg |
| 18:35.19 | [J]oules | yeah |
| 18:35.26 | [J]oules | just checked to be sure |
| 18:35.27 | Samot | -- Executing [18666556565@from-internal:1] Macro("PJSIP/290-00000181", "user-callerid,LIMIT,EXTERNAL,") in new stack |
| 18:35.28 | [TK]D-Fender | Not looking at a complete call. |
| 18:35.31 | Samot | This is an OUTBOUND CALL |
| 18:35.38 | Samot | Coming from an endpoint. |
| 18:35.41 | [J]oules | yes |
| 18:35.46 | Samot | Then going to another PBX |
| 18:35.50 | [J]oules | i dialed 8666556565 |
| 18:36.02 | [J]oules | pbx graciously adds 1 in front |
| 18:36.02 | Samot | Where is the INVITE from your PHONE? |
| 18:36.10 | [TK]D-Fender | We don't have it |
| 18:36.15 | [J]oules | no idea |
| 18:36.20 | Samot | OK. |
| 18:36.23 | [J]oules | i will do it again |
| 18:36.31 | [TK]D-Fender | "No idea"? Get a clue |
| 18:36.37 | Samot | goes back to real work. |
| 18:36.54 | [TK]D-Fender | goes to grab a coffee and go for a walk... |
| 18:37.23 | [J]oules | this is what i see from clean cli.... |
| 18:38.13 | *** join/#asterisk mike543 (404ac60a@gateway/web/freenode/ip.64.74.198.10) |
| 18:38.14 | [J]oules | https://www.irccloud.com/pastebin/OzIwWsv3/ |
| 18:38.31 | [J]oules | thats immediately after pressing redial on gs phone |
| 18:38.50 | [J]oules | i changed log settings perhaps thats why? |
| 18:39.54 | [J]oules | Samot: what should console have as default in log settings? |
| 18:40.03 | [TK]D-Fender | You posted incomplete debug |
| 18:40.04 | [TK]D-Fender | again |
| 18:40.05 | [TK]D-Fender | and again |
| 18:40.06 | [J]oules | turned off most because of moh streaming at night |
| 18:40.13 | [TK]D-Fender | Don't care |
| 18:40.16 | [J]oules | err lack of it |
| 18:40.19 | [TK]D-Fender | you wasted out time even looking at it |
| 18:40.27 | [J]oules | well something is preventing the invite from showing |
| 18:40.40 | [TK]D-Fender | Your lack of enabling the proper debug |
| 18:40.51 | [J]oules | wtf |
| 18:40.57 | [TK]D-Fender | there is no such thing as "preventing" only you not doing you job |
| 18:41.04 | [J]oules | sip set debug on; core set verbose 10 |
| 18:41.39 | [TK]D-Fender | facepalms |
| 18:42.16 | Samot | Is that how you typed it? |
| 18:42.47 | [TK]D-Fender | - Executing [18666556565@from-internal:1] Macro("PJSIP/290-00000186", "user-callerid,LIMIT,EXTERNAL,") in new stack |
| 18:42.55 | [TK]D-Fender | does that look like CHAN_SIP TO YOU? |
| 18:44.13 | [TK]D-Fender | https://www.irccloud.com/pastebin/UqozCcZf/ <- how about the start of THIS? |
| 18:44.33 | mike543 | hello, anyone know any tricks to send # via DTMF when using ARI ? I can send/make 0-9 tones but the # or anything past it in the string does not play. It works fine when done in the dialplan via senddtmf but not via ARI. I've tested with ARITESTER and curl but no love. Asterisk 13.1.0~dfsg-1.1ubuntu4.1. |
| 18:46.08 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 18:46.25 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Channels+REST+API#Asterisk15ChannelsRESTAPI-sendDTMF |
| 18:52.33 | mike543 | yes, when I do this I hear four tones POST http://localhost:8088/ari/channels/1527786693.36/dtmf?dtmf=1234 when I do this I hear one tone POST http://localhost:8088/ari/channels/1527786693.36/dtmf?dtmf=1#34 both respond with 204 No Content. |
| 18:54.04 | drmessano | host1.redacted-domain.tld <-- best part |
| 19:02.23 | [TK]D-Fender | goes for a walk... |
| 19:13.12 | [J]oules | https://www.irccloud.com/pastebin/Q84D27Rc/ |
| 19:13.33 | [J]oules | Samot: there are NO lines prior |
| 19:14.10 | Samot | I got a real support issue I'm working on. |
| 19:14.27 | Samot | You were already provided a huge red flag as to why you can't see your endpoint call. |
| 19:14.41 | drmessano | Maybe I should post my theory in here |
| 19:14.50 | drmessano | Since we know [J]oules has at least 3 personalities |
| 19:14.59 | drmessano | I also think he's Childish Gambino |
| 19:15.03 | drmessano | Thoughts? |
| 19:15.37 | drmessano | When have you ever seen [J]oules and Childish Gambino in the same room? Exactly |
| 19:16.06 | [TK]D-Fender | The logic is sound... |
| 19:17.11 | [J]oules | you mean freepbx log settings? |
| 19:17.24 | [J]oules | i put everything to on for console |
| 19:18.16 | mike543 | changing sip.conf dtmfmode from rfc2833 to inband didn't change the behavior. I hear four tones with 1234 but only 1 with 1#34 or none with #234. aha, if the string starts with # I get 400 Bad Request{ "message": "DTMF is required" } |
| 19:18.26 | [J]oules | https://usercontent.irccloud-cdn.com/file/k80YNqHx/fpbx-log-settings.png |
| 19:19.05 | [TK]D-Fender | meaningless |
| 19:19.24 | [J]oules | Samot: you wanted sip debug, sure you dont want pjsip logger? |
| 19:19.47 | Samot | JFC |
| 19:19.50 | [TK]D-Fender | YOU SAID YOU HAD A PJSIP AUDIO ISSUE WHY THE HELL ARE YOU ASKING IF YOU SHOULD ACTUALLY BE ***LOOKING*** AT THE CALL OR NOT? |
| 19:19.53 | Samot | SIP IS F'ING SIP |
| 19:20.13 | [J]oules | well ok, you see in that p/b exactly what I see |
| 19:20.14 | Samot | When we say "Provide a SIP Debug" we mean "Provide us with the SIP MESSAGES/TRANSACTIONS" |
| 19:20.29 | [TK]D-Fender | <[J]oules> well ok, you see in that p/b exactly what I see <- and you said NOTHING about looking at PJSIP debug |
| 19:20.34 | Samot | So use the PROPER SIP debugger |
| 19:20.44 | [TK]D-Fender | <[J]oules> sip set debug on; core set verbose 10 |
| 19:20.50 | [J]oules | yes |
| 19:20.50 | [TK]D-Fender | <[TK]D-Fender> - Executing [18666556565@from-internal:1] Macro("PJSIP/290-00000186", "user-callerid,LIMIT,EXTERNAL,") in new stack |
| 19:20.50 | [TK]D-Fender | <[TK]D-Fender> does that look like CHAN_SIP TO YOU? |
| 19:20.54 | Samot | Dude.. |
| 19:20.56 | Samot | Forget it. |
| 19:21.01 | [J]oules | ok |
| 19:21.03 | [J]oules | got it |
| 19:21.10 | [J]oules | you need both |
| 19:21.10 | Samot | He literally takes "sip debug" as the Chan_SIP debug command |
| 19:21.18 | Samot | Not as "SIP" in whole. |
| 19:21.19 | [J]oules | yup |
| 19:21.23 | [J]oules | you hit it right |
| 19:21.27 | Samot | Right. |
| 19:21.30 | Samot | Learn. |
| 19:21.32 | [TK]D-Fender | Idiot. The poison is in the OTHER half of the sandwich |
| 19:21.34 | [J]oules | ok got it |
| 19:21.56 | [J]oules | give me a few minutes and i will have both sip debug/pjsip logger |
| 19:22.02 | [J]oules | and a new call |
| 19:22.17 | [TK]D-Fender | That part of "look at the whole call" is still failing to sink in after YEARS of this nonsense? |
| 19:27.41 | [J]oules | isn't there supposed to be ulaw or g711 in this? |
| 19:27.55 | [J]oules | https://www.irccloud.com/pastebin/3NG9dsDC/ |
| 19:28.25 | *** join/#asterisk jeffspeff (~jeffspeff@209.141.208.197) |
| 19:28.29 | [TK]D-Fender | Why are you wasting our time with more partial garbage? |
| 19:29.09 | [J]oules | asking a freeking question |
| 19:29.21 | [TK]D-Fender | a STUPID one because don't see what's issuing that |
| 19:29.29 | [TK]D-Fender | And we don't have a reason to care |
| 19:29.45 | [TK]D-Fender | You configured your stuff. You are responsible |
| 19:31.01 | drmessano | lol |
| 19:31.13 | [J]oules | the phone has first 4 codecs setup pcmu |
| 19:31.23 | [J]oules | https://www.irccloud.com/pastebin/0Y5SdMU3/ |
| 19:31.46 | [J]oules | why is there no ulaw?? |
| 19:31.54 | [J]oules | that might be the problem right there |
| 19:32.15 | drmessano | JFC lol |
| 19:32.16 | [TK]D-Fender | a=rtpmap:0 PCMU/8000 <--- ULAW YOU TWAT |
| 19:32.43 | [J]oules | i am not a twat. |
| 19:32.49 | [TK]D-Fender | You're right |
| 19:33.04 | [TK]D-Fender | You have neither the depth, nor the warmth, and are completely undesirable.... |
| 19:33.06 | drmessano | You've been upgraded to MegaTwat |
| 19:33.16 | [TK]D-Fender | NEXT |
| 19:33.28 | [TK]D-Fender | And codec can't be the issue otherwise the call would have DROPPED |
| 19:33.44 | [TK]D-Fender | Even suspecting that is an idiotic waste of time showing you have no idea how SIP works |
| 19:33.56 | [TK]D-Fender | If it can't agree then it DIES INSTANTLY. |
| 19:34.01 | drmessano | I'm literally waiting for him to post something with the other IPv6 address in it so I can see if it's even pingable |
| 19:34.04 | [J]oules | ok ok let me finish getting call in p/b |
| 19:34.05 | drmessano | The first one is |
| 19:34.10 | [TK]D-Fender | <PROTECTED> |
| 19:34.21 | drmessano | I bet money the second one isn't |
| 19:34.41 | [TK]D-Fender | <[J]oules> ok ok let me finish getting call in p/b <- stop wasting our time with GARBAGE |
| 19:35.11 | *** part/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 19:35.11 | drmessano | He's lazy because he has to spend huge amounts of time sanitizing the logs |
| 19:35.14 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 19:35.18 | drmessano | He's lazy because he has to spend huge amounts of time sanitizing the logs |
| 19:35.36 | drmessano | I kinda just made that connection |
| 19:36.41 | [TK]D-Fender | drmessano, I don't see there being any need for pre-condition. Lazy is lazy, and incompetence prolific. |
| 19:37.21 | [TK]D-Fender | The latter supersedes all. |
| 19:37.22 | drmessano | No, i'm just saying, he's cutting corners due to his obsession with sanitizing his logs and the important information hidden in them. |
| 19:37.33 | *** join/#asterisk mducharme-laptop (~mducharme@remote.gobcn.ca) |
| 19:37.41 | drmessano | Which is why he argues "IM NOT LAZY" |
| 19:37.45 | [TK]D-Fender | "Let's micromanage things I can't read" |
| 19:37.49 | drmessano | No, he's wasting hours with log diddling |
| 19:38.00 | drmessano | Exactly |
| 19:38.06 | [TK]D-Fender | While continuing to NOT provide what was asked entire days prior |
| 19:38.24 | drmessano | Worried he's gonna provide more evidence that he's Lanny lol |
| 19:38.33 | [TK]D-Fender | "but here, let's look at this next context-free wild goose" |
| 19:39.22 | drmessano | TO give him some credit, imagine pulling hundreds of lines of debug, editing them in NOTEPAD, and then pasting them, only to find the debug is worthless anyway |
| 19:39.27 | drmessano | So back to square one |
| 19:39.32 | drmessano | Must be difficult lol |
| 19:40.20 | drmessano | "USELESS? BUT I SPENT THE LAST 2 HOURS.. SON OF A BUFFALA..." |
| 19:41.27 | [TK]D-Fender | "Difficult" isn't an option ... it comes bundled with the package. |
| 19:42.06 | drmessano | If he was smart, he would write a shell script to sed out all his domain name shit |
| 19:42.19 | drmessano | But I mean |
| 19:42.50 | drmessano | 1. If 2. He 3. Was 4. Smart |
| 19:43.00 | [TK]D-Fender | Or have a kayboard macro prepared for it for C?P |
| 19:43.11 | drmessano | That too |
| 19:43.41 | drmessano | Maybe Lanny can help him with that |
| 19:43.46 | drmessano | Since they're such good buddies |
| 19:44.30 | [J]oules | http://pastebin.ca/4034405 |
| 19:47.16 | [TK]D-Fender | waits for the other half of the call... |
| 19:48.22 | drmessano | He masked the IP |
| 19:48.24 | drmessano | lol |
| 19:48.31 | [J]oules | what other half? |
| 19:48.37 | drmessano | I'm guessing this host ---> [xxxx:xxxx:1001::b154:4edb] is unreachable |
| 19:49.03 | drmessano | I never see anything from it in the debug |
| 19:49.05 | [J]oules | host2 server? |
| 19:49.23 | drmessano | So it's probably the misconfigured one |
| 19:52.50 | mducharme-laptop | the log is dumped from the [xxxx:xxxx:1001::b154:4edb] server |
| 19:53.04 | mducharme-laptop | he says |
| 19:53.34 | drmessano | Can he ping6 both ends? |
| 19:53.41 | drmessano | Host <> Host ? |
| 19:54.25 | [J]oules | yes |
| 19:54.50 | mducharme-laptop | there are firewall rules on the routers on both sides to allow all tcp and udp traffic to and from that IP |
| 19:55.39 | mducharme-laptop | no ports are being blocked, so everything should make it through to that server and vice versa |
| 19:57.23 | Samot | OK. |
| 19:57.27 | Samot | Let's just cut the crap. |
| 19:57.27 | [J]oules | mducharme-laptop: the audio does make it back from host2 to the 4edb pbx |
| 19:57.42 | Samot | This TLS/PJSIP/IPv6 setup has been a nightmare for him. |
| 19:57.44 | Samot | And for us. |
| 19:57.54 | Samot | Almost every other week it is something new. |
| 19:57.56 | [J]oules | i have the recording of the calls that I cannot hear, but the recording heard it |
| 19:58.14 | Samot | Because the issue is in between the PBX and the endpoint. |
| 19:58.15 | Samot | Done. |
| 19:58.21 | [J]oules | playback of the calls with no audio has audio |
| 19:58.25 | Samot | It's a networking problem or something that involves your configs. |
| 19:58.28 | Samot | Go look and fix them. |
| 19:58.30 | [J]oules | i agree with Samot |
| 19:58.48 | Samot | I highly suggest you stop using IPv6 |
| 19:58.52 | [J]oules | if the recording was blank that would be a different story |
| 19:59.09 | drmessano | I think the issue is specifically something with the IPv6 implementation |
| 19:59.10 | [J]oules | well what is wrong with ipv6? |
| 19:59.16 | drmessano | Maybe a provider, maybe a config |
| 19:59.19 | [J]oules | its the future |
| 19:59.22 | Samot | Sure. |
| 19:59.25 | drmessano | Your setup sucks |
| 19:59.30 | drmessano | or your provider sucks |
| 19:59.34 | Samot | It's a config or setup issue. |
| 19:59.36 | drmessano | IPv6 isn't the problem |
| 19:59.47 | [J]oules | in firewall? |
| 19:59.50 | drmessano | It's the implementation |
| 19:59.51 | [J]oules | in MT firewall? |
| 19:59.56 | Samot | Go look |
| 20:00.00 | Samot | Dude, just stop. |
| 20:00.05 | Samot | Stop wasting time in here. |
| 20:00.11 | drmessano | IPv6 is not a NAT fixer |
| 20:01.30 | drmessano | Empirical evidence time... |
| 20:01.45 | drmessano | He implemented IPv6 because IPv4+NAT was giving him fits |
| 20:01.53 | drmessano | Which is a 2007 problem, not a 2018 problem |
| 20:02.01 | drmessano | It's not that hard |
| 20:02.06 | drmessano | and we never got an explanation |
| 20:02.18 | drmessano | So IPv6 was thrown into the mix, and it breaks, repeatedly |
| 20:02.32 | drmessano | That tells me, it's not the choice of IP Stack or NAT |
| 20:02.51 | Samot | It's not. |
| 20:02.54 | mducharme-laptop | I looked at his firewall - on the side where he is, and the phones, rules to allow all tcp and udp from the sip server ip address.. on the router at the sip server side, rules to allow from anywhere ports 5060, 5061, 5161 tcp and udp, and 10000-20000 udp only |
| 20:03.19 | Samot | EXCEPT |
| 20:03.32 | Samot | The phones don't use those ranges. |
| 20:03.50 | Samot | And has he mucked with the PBX and changed the RTP range? |
| 20:04.46 | mducharme-laptop | I don't know if he has or not, he told me that the pbx used 10000 to 20000 so that's what was entered on the destination port side of the rule to allow the traffic through |
| 20:04.59 | Samot | Well.. |
| 20:05.06 | Samot | Since this is a few month old release.. |
| 20:05.06 | mducharme-laptop | if the pbx is using other rtp ports then that isn't going to work without adjusting the rules |
| 20:05.20 | Samot | Are there any issues in the issue tracker that address this problem? |
| 20:06.16 | mducharme-laptop | I haven't checked, I don't really know the asterisk side very well, I was just checking out his ipv6 routing and firewall config |
| 20:06.42 | Samot | If 13.19.1 has a bug that breaks RTP over IPv6/PJSIP it would be in the tracker. |
| 20:06.42 | mducharme-laptop | he would have to answer that |
| 20:07.19 | Samot | Changelogs, etc.. |
| 20:07.22 | Samot | All that would be there.. |
| 20:07.28 | Samot | It would take like 10 minutes to look at. |
| 20:07.44 | Samot | Since the claim is "I updated to 13.19.1 and things are now broken" |
| 20:07.56 | Samot | Well, the current release is 13.21.0. |
| 20:08.04 | Samot | Between those releases were there issues reported? |
| 20:09.55 | mducharme-laptop | I'll try creating a rule to allow all traffic from the range the phone handsets are on, in case the traffic is using some port outside of taht range |
| 20:10.43 | Samot | If the issue is the endpoint isn't getting audio but the audio exists at the PBX |
| 20:10.56 | Samot | Then the issue is between the RTP leaving the PBX and getting to the phone. |
| 20:11.11 | Samot | Is the RTP being sent to the proper IP? |
| 20:13.46 | [TK]D-Fender | <[J]oules> what other half? <- you're calling another one of YOUR servers. a whoel stack of other garbage that can go wrong |
| 20:14.29 | [TK]D-Fender | And the test circumstances are STUPID for complicating it with 10 other things being added |
| 20:14.43 | Samot | Which PBX was this call recorded on? |
| 20:14.52 | Samot | The billing PBX or the client PBX? |
| 20:15.05 | [TK]D-Fender | or MULTIPLE other servers |
| 20:15.09 | drmessano | I have a dumb question |
| 20:15.18 | [TK]D-Fender | again we only see one hop in whatever this happens to be |
| 20:15.22 | drmessano | Where is the Polycom that doesn't work? |
| 20:15.24 | drmessano | Home? |
| 20:15.36 | drmessano | Have we pinged HOME <> PBX and PBX <> HOME |
| 20:15.36 | [TK]D-Fender | drmessano, It's magically a GS now... |
| 20:15.43 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 20:15.45 | drmessano | Have we ping6ed HOME <> PBX and PBX <> HOME |
| 20:16.10 | [TK]D-Fender | drmessano, We've seen nothing sensible for debugging at all |
| 20:16.18 | drmessano | Have we ping6ed $LocationOfIPPhone <> PBX and PBX <> $LocationOfIPPhone |
| 20:17.53 | Samot | 1:39:17 PM <[J]oules> confirmed polycom set to 'prefer ipv6' has no audio on calls. Rebooting it now with network setting to ipv4 only |
| 20:17.53 | Samot | User-Agent: Grandstream GXP2130 1.0.9.69 |
| 20:18.05 | Samot | Yup, the goal posts have changed. |
| 20:18.07 | drmessano | What is :fe95:223a ? |
| 20:18.19 | [TK]D-Fender | This is all just senseless.... |
| 20:18.21 | Samot | Now we're testing with a device that's not even part of the original issue stated. |
| 20:18.50 | drmessano | I still don't see that IPv6 has been proven between endpoint and PBX |
| 20:18.58 | mducharme-laptop | the sip server can ping the phone, and the router where the phone is can ping the sip server |
| 20:18.59 | [TK]D-Fender | Information is a lie. Debug incomplete. Testing methodology is stupid altogether. |
| 20:19.05 | drmessano | Okasy |
| 20:19.08 | drmessano | Okay |
| 20:19.26 | [TK]D-Fender | Anybody who works like this shouldn't be in the field. |
| 20:20.37 | mducharme-laptop | the firewall on the router at both the handset side and the pbx side are configured in such a way that nothing should get blocked between the two |
| 20:20.54 | [TK]D-Fender | #hearsay |
| 20:21.07 | mducharme-laptop | hmm? |
| 20:21.21 | [TK]D-Fender | We have assurances, not proof. |
| 20:21.27 | drmessano | I trust mducharme-laptop on this.. |
| 20:21.37 | drmessano | If he's looking at and configured the firewall |
| 20:21.48 | *** join/#asterisk miralin (~Thunderbi@91.237.94.7) |
| 20:21.53 | drmessano | So we're back to IPv6 config on the PBX |
| 20:21.59 | drmessano | or Asterisk |
| 20:22.26 | drmessano | So now what about a ping from the router where the endpoint resides to the PBX? |
| 20:22.36 | drmessano | and from the PBX to the endpoints router? |
| 20:22.38 | [TK]D-Fender | Anything coming from Secret Squirrel's system get an automatic "extreme prejudice" flag on it. |
| 20:23.13 | mducharme-laptop | yup, the ping from the router at the endpoint to the pbx works |
| 20:23.40 | mducharme-laptop | the ping from the pbx to the endpoint works and the ping from the pbx to the endpoint's router will also work |
| 20:23.56 | mducharme-laptop | although you can get [J]oules to try that too, I already know it will work |
| 20:24.33 | drmessano | PBX to the endpoints router will confirm the IPv6 stack on the PBX |
| 20:25.34 | drmessano | and what about my iptables stuff that he's butchered endlessly and put on his boxes? Has he killed iptables on the box? |
| 20:27.05 | mducharme-laptop | he was only running iptables and not ip6tables, the router is protecting the pbx when it comes to v6 |
| 20:27.12 | drmessano | ok |
| 20:27.22 | mducharme-laptop | sip1 can ping the ip of the router where the endpoints are |
| 20:29.41 | Samot | 4:14:42 PM <Samot> Which PBX was this call recorded on? |
| 20:32.01 | [J]oules | Samot: the freepbx my phone connects to |
| 20:49.03 | mike543 | aha, the http request is dropping everything after the #. this is in my curl command: POST "http://localhost:8088/ari/channels/1527799505.68/dtmf?dtmf=1#34" but I see this in curl output: > POST /ari/channels/1527799505.68/dtmf?dtmf=1 HTTP/1.1 |
| 20:52.49 | mducharme-laptop | ok so I'm going to set up a rule on the router at the pbx side to capture traffic to and from that phone |
| 20:52.52 | mducharme-laptop | to log |
| 20:53.34 | mike543 | woohoo!!! http://localhost:8088/ari/channels/1527799505.68/dtmf?dtmf=%23 |
| 20:55.28 | *** join/#asterisk Oatmeal (Suzeanne@gateway/vpn/privateinternetaccess/suzeanne) |
| 20:56.25 | mducharme-laptop | ok so |
| 20:56.53 | mducharme-laptop | when he makes a call from the handset, the traffic hits the pbx router and heads toward the pbx, udp port 11256 |
| 20:57.21 | mducharme-laptop | nothing comes back from the pbx to the handset except icmp |
| 21:01.16 | mducharme-laptop | and the sip replies |
| 21:01.31 | mducharme-laptop | but RTP is only going one way, from the handset to the pbx |
| 21:12.45 | Samot | Well.. |
| 21:12.54 | Samot | That is a rather vague presumption. |
| 21:13.04 | Samot | More like the RTP isn't hitting the router. |
| 21:13.23 | Samot | He could run an RTP debug on the system. |
| 21:13.41 | Samot | If it's not "sending" RTP there will be no outgoing RTP packets in the debug. |
| 21:13.50 | Samot | If it's sending RTP it will show where it's sending it. |
| 21:14.55 | Samot | mducharme-laptop: This isn't the first time we've had to deal with this type of issue with him. He has been provided commands and instructions numerous times previously for troubleshooting these issues. |
| 21:15.13 | Samot | The fact we have to re-teach these steps each time is rather annoying. |
| 21:18.18 | mducharme-laptop | well what I do know is that the RTP is making it from the handset to the router where the PBX is, whether it is making it to the PBX from that router I don't know, and whether the PBX is replying I don't know, but if it is, the replies are not making it back to the router where the PBX is |
| 21:18.42 | Samot | And there are tools to determine this. |
| 21:21.43 | Samot | mducharme-laptop: If you are relying on [J]oules to provide you with details or know what to do or what steps to take to fill in the gaps of your knowledge when it comes to Asterisk. You are relying on the wrong thing. You should know this by now. |
| 21:22.35 | Samot | Part of the issue here is, you don't know one side of the equation. So you have no idea what to look at to confirm or test any of this stuff from the Asterisk side. [J]oules is just as un-knowledgeable as you. |
| 21:23.34 | Samot | So at this point I don't think it's an issue with the routers, because you have looked and made changes and we trust that 100%. |
| 21:27.09 | mducharme-laptop | yeah, I haven't had to do very much asterisk troubleshooting, I do admin stuff on our pbx but it has the phones on the same subnet as it and no trunks to the outside, so not a lot can go wrong |
| 21:27.39 | mducharme-laptop | actually the only thing that went wrong is when I was making some changes and fail2ban decided to block all of the handsets |
| 21:27.56 | mducharme-laptop | took a half hour to figure out what had happened |
| 21:28.27 | mducharme-laptop | disabled fail2ban but it restarted again automatically on the next reboot (freepbx distro) |
| 21:28.36 | mducharme-laptop | so I just wound up whitelisting the subnet |
| 21:28.51 | mducharme-laptop | it's really stupid though, don't see why I need fail2ban there |
| 21:32.00 | [J]oules | Samot what 'other' side are you referring to?? |
| 21:34.34 | Samot | The side that requires someone to understand how Asterisk works. |
| 21:36.02 | Samot | mducharme-laptop says the routers are setup and configured properly for this, then we're good with that. As long as all the right information was provided for him to do so. |
| 21:36.56 | Samot | However, to determine if Asterisk is doing things right or to troubleshoot on the Asterisk side. His knowledge is limited which means someone has to have that knowledge to make sure the Asterisk is correct or to troubleshoot that side. |
| 21:39.39 | Samot | Honestly, at this point if you have mducharme-laptop working on the routers and fixing problems you might has well dump IPv6 and fix whatever NAT woes were haunting you no IPv4. |
| 21:39.44 | Samot | s/no/on/ |
| 21:40.02 | Samot | Because you'll just have less problems. |
| 21:40.41 | Samot | IPv6 has been "the future" for 20+ years and most SIP devices still have basic or poor IPv6 support/implementation. |
| 21:40.51 | Samot | Including B2BUAs. |
| 21:43.32 | Samot | We are pretty much aware that your deployment of IPv6 was nothing more than a "fix" to your NAT issues. |
| 21:49.05 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:51.50 | [J]oules | We don't have any NAT issues |
| 21:57.45 | [J]oules | So I take it from that very large sip debug, nothing could be found that would tell you why there's no audio on the phone |
| 21:59.24 | Samot | I said what to do |
| 21:59.56 | Samot | Do a rtp debug to see if and where the rtp goes |
| 22:09.34 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:31.10 | [J]oules | i did do that |
| 22:31.19 | [J]oules | going to the right ipv6 address |
| 22:31.54 | [J]oules | its not just gs phone, tried on polycom and same thing, no audio |
| 22:32.47 | *** join/#asterisk sibyakin (~sibyakin@188.162.228.243) |
| 22:35.17 | *** join/#asterisk rpifan (~rpifan@2600:1:c151:e382:905c:a615:7add:5974) |
| 22:45.48 | Samot | And are these phones directly on the router? Or do they connect to a second device? |
| 22:46.15 | [TK]D-Fender | Did we ever get a direct non-polluted test? |
| 22:46.50 | Samot | What do you think? |
| 22:47.07 | mducharme-laptop | [J]oules did the rtp debug on the asterisk box, it shows it sending the packet to the handset at the correct address |
| 22:47.16 | mducharme-laptop | however that packet does not make it to the router |
| 22:47.20 | mducharme-laptop | the near router |
| 22:47.53 | Samot | The router at the DC or the router at the house/whereever. |
| 22:48.20 | mducharme-laptop | correct, the server is at the dc and it doesn't even make it to the router at the DC, let alone the router where the endpoints are |
| 22:48.42 | Samot | Did anyone do a tcpdump at the server level? |
| 22:48.52 | Samot | Asterisk shows the packets leaving Asterisk. |
| 22:49.04 | Samot | Is there UCP traffic at the interface level? |
| 22:49.06 | mducharme-laptop | that's actually what I asked him to do just now |
| 22:49.29 | Samot | It should have been in the list of the first 6 things. |
| 22:49.59 | mducharme-laptop | because either it is being logged like it is being sent out but actually isn't, or it is getting blocked between the server and the router |
| 22:50.04 | *** join/#asterisk rpifan (~rpifan@2600:1:c151:e382:905c:a615:7add:5974) |
| 22:50.07 | Samot | Does this happen on inbound calls? |
| 22:50.20 | Samot | When the PBX originates the call to the endpoint? |
| 22:50.22 | [TK]D-Fender | You mean ... like something maybe a direct firewall on the server might kill? |
| 22:50.39 | mducharme-laptop | there are two possibilities, firewall on the server or on the kvm host |
| 22:50.44 | Samot | Does this happen on inbound calls? |
| 22:50.54 | Samot | When the PBX originates the call to the endpoint? |
| 22:51.25 | mducharme-laptop | don't know, you would have to ask [J]oules |
| 22:51.32 | Samot | I am. |
| 22:53.03 | Samot | 6:50:20 PM <[TK]D-Fender> You mean ... like something maybe a direct firewall on the server might kill? |
| 22:53.21 | Samot | ^^ Either that or the traffic isn't being treated right. |
| 22:53.35 | Samot | At a higher level. |
| 22:54.44 | [J]oules | origination call has no audio either |
| 22:55.02 | mducharme-laptop | did you do the tcpdump? |
| 22:55.07 | [J]oules | even sip uri origination doesnt have audio |
| 22:55.07 | Samot | And iptables is completely disabled? |
| 22:55.09 | [J]oules | yeah |
| 22:55.17 | [J]oules | iptables is up and running |
| 22:55.25 | Samot | WTF |
| 22:55.27 | Samot | TURN IT OFF |
| 22:55.29 | mducharme-laptop | but ip6tables is turned off |
| 22:55.30 | Samot | Then TEST |
| 22:55.30 | [J]oules | ok |
| 22:55.44 | mducharme-laptop | iptables is for ipv4 only in general |
| 22:55.52 | *** join/#asterisk rpifan (~rpifan@2600:1:d10c:dfeb:905c:a615:7add:5974) |
| 22:55.55 | Samot | It's [J]oules. |
| 22:56.04 | Samot | So the answer is, turn if off and then test. |
| 22:56.06 | [J]oules | wtf are you saying now Samot |
| 22:56.12 | Samot | Just turn it off and test. |
| 22:56.14 | Samot | GD. |
| 22:56.26 | [J]oules | its [J]oules |
| 22:56.30 | [J]oules | wtf is that |
| 22:56.33 | Samot | This is a 30 minute problem. |
| 22:56.39 | [TK]D-Fender | Turn that shit the fuck >>> OFF <<< |
| 22:56.39 | Samot | That's taken almost 24 hours. |
| 22:56.41 | [J]oules | really? |
| 22:56.43 | Samot | Just turn it off and TEST |
| 22:56.46 | Samot | YES |
| 22:56.48 | [J]oules | OK GD |
| 22:56.55 | [TK]D-Fender | You're not getting audio and you leave shit in the way |
| 22:57.17 | [J]oules | ahhh centos 7 and iptables stop do not get along |
| 22:57.42 | [J]oules | cant turn it off |
| 22:57.48 | Samot | Dear god. |
| 22:57.51 | [J]oules | it has its own mind |
| 22:57.58 | [J]oules | service iptables stop |
| 22:58.06 | [TK]D-Fender | At least that makes one of you.... |
| 22:58.12 | mducharme-laptop | wouldn't it be systemctl stop iptables? |
| 22:58.14 | [J]oules | # service iptables stop |
| 22:58.14 | [J]oules | Redirecting to /bin/systemctl stop iptables.service |
| 22:58.45 | [J]oules | ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5061 |
| 22:58.49 | mducharme-laptop | you can just run systemctl stop iptables in centos 7, not that it shoul make a difference |
| 22:59.10 | Samot | What did the tcpdump show? |
| 22:59.12 | [TK]D-Fender | [J]oules> ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5061 <- meaningless |
| 22:59.17 | Samot | Is there UDP traffic for the RTP ports? |
| 22:59.34 | [J]oules | i am just unzipping and now wireshark decided to upgrade |
| 22:59.39 | Samot | Leaving the server going to the IP. |
| 22:59.39 | [J]oules | so give me a sec |
| 22:59.47 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |
| 23:00.32 | [J]oules | Got RTP packet from xxx.xx.49.40:11512 (type 00, seq 016504, ts 2344175736, len 000160) |
| 23:00.32 | [J]oules | Sent RTP packet to [2607:fea8:20df:fc31:20b:82ff:fe95:223a]:5008 (type 00, seq 008043, ts 2344175736, len -000013) |
| 23:00.45 | [J]oules | that ipv6 address is my grandstream phone |
| 23:00.45 | Samot | That's the RTP debug. |
| 23:00.49 | [J]oules | yes |
| 23:00.53 | Samot | I asked what the tcpdump showed. |
| 23:00.56 | [J]oules | just to be clear |
| 23:01.04 | [J]oules | well hang on Mr. Samot |
| 23:02.09 | [TK]D-Fender | <[J]oules> Got RTP packet from xxx.xx.49.40:11512 (type 00, seq 016504, ts 2344175736, len 000160) |
| 23:02.09 | [TK]D-Fender | <[J]oules> Sent RTP packet to [2607:fea8:20df:fc31:20b:82ff:fe95:223a]:5008 (type 00, seq 008043, ts 2344175736, len -000013) |
| 23:02.16 | [TK]D-Fender | MIXED IPV4 IPV6 |
| 23:03.21 | Samot | Only the endpoints talk IPv6 |
| 23:03.23 | Samot | The trunks don't. |
| 23:04.34 | [TK]D-Fender | We shouldn't even be using a trunk |
| 23:04.42 | mducharme-laptop | sorry if this is a stupid question b/c I'm not familiar with some of the details of asterisk, but is it the exact same RTP stack regardless of whether chan_sip or pjsip is used? |
| 23:04.51 | [TK]D-Fender | if he's blaming the comms with the phone he shouldn't have that other crap polluting the test |
| 23:06.02 | Samot | I believe so. |
| 23:06.35 | drmessano | Stupid question: the FQDN of the system doesnât match the external hostname again, does it? |
| 23:06.35 | mducharme-laptop | I tried calling him from a test extension that he set up for me on the same box - do you have a debug for that call [J]oules? |
| 23:07.02 | Samot | Did it have the same issue? |
| 23:07.07 | mducharme-laptop | yes |
| 23:07.13 | Samot | How about we just see what the hell this tcpdump says |
| 23:07.20 | Samot | Before we start looking at other stuff. |
| 23:07.24 | mducharme-laptop | yes I think we need to see the tcpdump |
| 23:07.25 | drmessano | Because you know the IP6 entries in a host file are different |
| 23:07.31 | Samot | You said you were opening it |
| 23:07.35 | Samot | I'm just waiting.. |
| 23:07.41 | [TK]D-Fender | Samot> How about we just see what the hell this tcpdump says <- can't argue that.... |
| 23:07.46 | drmessano | He could be literally replying back to himself |
| 23:08.06 | mducharme-laptop | I'm going to look at it |
| 23:08.19 | mducharme-laptop | because I can compare with the packet logging I'm doing on the router |
| 23:08.33 | Samot | All I want to know... |
| 23:08.42 | Samot | Since Asterisk is showing the RTP leaving Asterisk.. |
| 23:08.56 | Samot | Is it hitting the ETHER port and going out? |
| 23:09.08 | Samot | Because the claim is, the router in front of the PBX never gets it. |
| 23:09.25 | mducharme-laptop | that's what I want to know as well |
| 23:09.26 | drmessano | Cough */etc/hosts* |
| 23:09.52 | Samot | ^^ check that too. |
| 23:09.55 | mducharme-laptop | if the packet is in tcpdump but doesn't reach the router, the problem has to be in KVM |
| 23:10.04 | Samot | Well... |
| 23:10.06 | [TK]D-Fender | drmessano, Classic fail for many using dyndns.... |
| 23:10.15 | drmessano | Heâs done that shit with the external host name in the host file several times |
| 23:10.17 | Samot | [J]oules: How did you update this box? |
| 23:10.37 | drmessano | It wouldnât surprise me if heâs only broken it for IPv6 |
| 23:11.01 | mducharme-laptop | and it is actually possible the problem is in KVM |
| 23:11.10 | Samot | Well if Asterisk was updated by asterisk-version-switch then there's other issues. |
| 23:11.23 | Samot | If the box was updated with a system update, the /etc/hosts could have been re-written. |
| 23:11.29 | Samot | Because I've had that happen |
| 23:11.36 | Samot | So what drmessano is saying, should be checked. |
| 23:12.26 | drmessano | He likes to name the boxes the same thing as the external host name, and then to fix what breaks he removes entries from the host file⦠If theyâve been put back itâs probably screwed up the box |
| 23:12.48 | drmessano | This is a repeat offense |
| 23:14.07 | mducharme-laptop | hmm |
| 23:14.15 | Samot | 24 hours for 30 minutes of troubleshooting. |
| 23:14.21 | Samot | TOPS |
| 23:14.37 | mducharme-laptop | may I ask why it woud be a problem to name the box the same as the external host name? |
| 23:15.01 | mducharme-laptop | er would |
| 23:16.02 | mducharme-laptop | I'm curious about that, because maybe there is something that I don't quite understand |
| 23:16.13 | drmessano | Because you can shove an external host into asterisk and it will resolve the DNS name and use the resultant IP address in the invites⦠|
| 23:16.30 | mducharme-laptop | and that is a problem because..? |
| 23:16.45 | drmessano | The box canât reach 127.0.0.1 |
| 23:17.04 | drmessano | Well other devices I mean |
| 23:17.09 | mducharme-laptop | ahh |
| 23:17.22 | rpifan | asterisk i love thee |
| 23:17.24 | mducharme-laptop | so you are saying that it adds a record in the hosts file for 127.0.0.1 and the hostname of the box |
| 23:17.25 | rpifan | but i have no projects for it |
| 23:18.04 | drmessano | And I configure the externhost to be the external host name, itâs going to resolve that from the hosts file to 127.0.0.1 |
| 23:18.09 | drmessano | Heâs done it a few times |
| 23:18.38 | drmessano | And again, it obviously isnât broken for IP4, but I donât know what the IP6 config looks like on the box |
| 23:21.27 | drmessano | Also, what was the prior asterisk version? |
| 23:24.04 | drmessano | Because, guess what, chan_pjsip only recently supported the chan_sip equivalent of externhost resolving to a hostname.. before it needed an IP address |
| 23:25.12 | drmessano | so if he has an FQDN in Asterisk and it's NOW resolving it after the upgrade, the FQDN resolves to ::1... BOOM |
| 23:25.33 | drmessano | Just a thought |
| 23:25.42 | drmessano | and explaining the "why thats bad" |
| 23:27.55 | mducharme-laptop | ok I understand the problem |
| 23:28.02 | mducharme-laptop | but what is the reason for using externhost to begin with? |
| 23:28.22 | mducharme-laptop | or do you have to |
| 23:28.30 | Samot | You don't have to |
| 23:28.53 | drmessano | Sigh |
| 23:28.57 | drmessano | You dont have to |
| 23:29.07 | drmessano | But most plug that in because FreePBX ASKS FOR IT |
| 23:29.17 | Samot | ^^^ |
| 23:29.20 | Samot | It's a box, in a form. |
| 23:29.22 | Samot | Fill it out |
| 23:29.23 | drmessano | and when the box is named the same as the external hostname, boom |
| 23:29.25 | mducharme-laptop | ok, got it |
| 23:30.10 | drmessano | Side effect of it being a neat tool for DDNS users, having Asterisk resolve the hostname and use it in INVITES, is that it's a common point of breakage for some |
| 23:30.46 | drmessano | They name the box sip.mydomain.com and plug that into the idiot box in the Web UI |
| 23:30.55 | drmessano | So boom, loopback |
| 23:31.08 | mducharme-laptop | right |
| 23:31.45 | drmessano | Also, our friend has done this multiple times.. and every couple years he breaks it and forgets why he was told it didnt work |
| 23:31.58 | drmessano | Which is why I suggested checking it, at least for IPv6 |
| 23:32.55 | drmessano | 13.19.0 was the first version to support this with PJSIP |
| 23:33.01 | drmessano | ding ding ding |
| 23:34.23 | drmessano | mducharme-laptop: So check our friends box.. he has me on ignore, so he's reading none of this |
| 23:34.55 | mducharme-laptop | I'm trying to, but he can't seem to figure out how to set up an account on ftp to download the pcap file |
| 23:35.06 | mducharme-laptop | er trying to check the packet capture that is |
| 23:35.44 | Samot | How about you just have him pastebin you privately the /etc/hosts file so you can check that |
| 23:35.54 | drmessano | ^^^^ Yes please |
| 23:43.33 | drmessano | I cant hang any longer.. If it's the hosts file, ping me so I get some validation |
| 23:43.52 | drmessano | But everything adds up empirically.. just need the actual proof |
| 23:44.11 | mducharme-laptop | he has vanished |
| 23:44.40 | mducharme-laptop | waiting for him to respond |
| 23:46.04 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 23:48.10 | drmessano | Yep, sounds right |
| 23:49.03 | drmessano | Iâll try not to say something about all the crap I get for my responses to him, and though they are a bit exaggerated, they are not baseless |
| 23:49.10 | drmessano | Going down for a nap |
| 23:49.29 | mducharme-laptop | he's getting the hosts file now |
| 23:51.31 | mducharme-laptop | the hosts file has localhost and localhost.localdomain for 127.0.0.1 and ::1 |
| 23:51.45 | drmessano | Ok and? |
| 23:51.50 | mducharme-laptop | it has the public ipv4 and public ipv6 for the regular dns name |
| 23:51.59 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:52.15 | drmessano | Which doesnât need to be in there at all |
| 23:59.01 | mducharme-laptop | does it cause a problem if it is there? |
| 23:59.29 | mducharme-laptop | I imagine maybe it gets written there by default |
