| 00:10.09 | *** part/#asterisk kharwell (kharwell@nat/digium/x-nshbhbzqyddgfezb) |
| 00:28.19 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
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| 00:39.19 | Kobaz | [2017-11-20 12:25:09.991] WARNING[7309] chan_sip.c: Autodestruct on dialog '1926a28b52eefd9d4e3af4c740e23a8a@192.168.50.1:5060' with owner SIP/104-00000ce3 in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 00:39.22 | Kobaz | what would cause that? |
| 00:50.07 | Samot | Are you running something in the h extension? |
| 00:50.25 | Samot | That means there's something happening with a call that is no longer happening. |
| 00:52.49 | *** join/#asterisk Korolev (~Korolev@c-73-179-85-246.hsd1.fl.comcast.net) |
| 00:53.07 | Kobaz | right |
| 00:53.08 | Kobaz | hah |
| 00:53.15 | Kobaz | trying to find more.. exactly why |
| 00:53.23 | Samot | Well, are you doing something with the calls at hangup? |
| 00:53.31 | Samot | Are you running a query? Calling a script? |
| 00:53.36 | Kobaz | yeah, but, this is happening before the hangup |
| 00:53.41 | Kobaz | this is the *cause* of the hangup |
| 00:54.04 | Samot | That means it got a BYE. |
| 00:54.15 | Samot | Show a full call this happens on. |
| 00:54.32 | Kobaz | okay so it's receiving a BYE, not sending a BYE due to some issue |
| 00:54.41 | Samot | I don't know. |
| 00:54.42 | Samot | Show a full call this happens on. |
| 00:54.45 | Kobaz | i don't have the sip capture on this, so i don't have the exact sequencing |
| 00:54.56 | Samot | Need to see it happen |
| 00:54.59 | Kobaz | right |
| 00:55.19 | Kobaz | i need to get homer up one of these days |
| 00:55.28 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 00:55.54 | Kobaz | https://pastebin.ca/3937909 |
| 00:56.30 | Kobaz | there's really no reason not to continually capture all non register/options sip dialogs |
| 00:56.51 | Kobaz | unless you're handling like 100 calls a second, it would be pretty neglegable load |
| 00:57.20 | Kobaz | honestly it looks like something bad happened internally, and asterisk is sending a BYE |
| 00:57.28 | Kobaz | i gotta look though the code to see |
| 00:58.07 | Kobaz | i've seen this before on my own box in the office... calls randomly drop apparently for no reason |
| 00:58.19 | Kobaz | i have captured those... and asterisk sends a BYE during setup |
| 00:58.37 | Samot | Need to see a real debug |
| 00:58.44 | Samot | Need to see the BYE messages |
| 00:58.45 | Kobaz | that's all i got |
| 00:58.47 | Kobaz | right |
| 00:58.54 | Kobaz | i told you i don't have the sip dialog |
| 01:01.44 | Kobaz | so basically, that log item is from sip_scheddestroy() |
| 01:01.52 | Kobaz | and it's called from a variety of places |
| 01:02.33 | Kobaz | notice the softhangup is after the autodestruct |
| 01:02.43 | Kobaz | so autodestruct caused the softhangup |
| 01:02.55 | Oeaa | Is it pretty normal to have a log completely spammed full of jitterbuffer resyncs? |
| 01:02.55 | Kobaz | if it was a normal hangup, you would see softhangup first |
| 01:03.02 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 01:03.19 | Kobaz | Samot: oej's whodoneit patch would be useful here |
| 01:03.36 | Kobaz | i forgot the name, he's got some crazy names... pinefrog and whatnot |
| 01:04.24 | Oeaa | [Nov 20 18:00:56] WARNING[20059][C-00000002] chan_iax2.c: Resyncing the jb. last_delay 5, this delay -47385, threshold 1050, new offset 56479 |
| 01:04.34 | Kobaz | oh, iax |
| 01:04.51 | Oeaa | just completely packed full, the log file... every several seconds. |
| 01:05.19 | Oeaa | I've had several agents with some pretty bad stuttering but I wrote it up as packetloss, netstats revealed as much |
| 01:05.27 | Oeaa | but I was just wondering if i'm missing something here with a log file full of that.. |
| 01:06.47 | Kobaz | Oeaa: i'm checking out the logging in the module for this... there's no verbosity setting for that message |
| 01:06.56 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 01:07.10 | Kobaz | so that message is printed when there's a large change in delay |
| 01:07.34 | Oeaa | delay as in latency? because it's just completely spammed... but I also have around 30 agents on and dialing |
| 01:07.53 | Kobaz | i don't really know the jitterbuffer code |
| 01:07.58 | Kobaz | that's a good #asterisk-dev question |
| 01:08.09 | Oeaa | yeah me either, I may just turn off the adaptive jitterbuffer and see how performance is. |
| 01:08.17 | Oeaa | Maybe switch everyone over to speex from u-law |
| 01:08.40 | Kobaz | <PROTECTED> |
| 01:08.48 | Kobaz | that's basically it for the docs on that, so far |
| 01:09.08 | Kobaz | it sounds like you have a lot of jitter, and the swing is wide |
| 01:09.29 | Kobaz | Oeaa: do an mtr on some of your endpoints that seems to have high latency |
| 01:10.27 | Oeaa | Hmm not a bad idea. PBX running in a vmware esxi as well so i'm sure it's not the most ideal situation |
| 01:10.41 | Oeaa | i'm sure there's some timing issues |
| 01:10.52 | Kobaz | how loaded down is the host? |
| 01:11.27 | Oeaa | the vm.. not so much, the host.. i honestly couldn't tell you because I don't have any access to it. Just have access to the PBX itself. |
| 01:11.45 | Oeaa | VM sits at about 27% cpu util with 30 or so agents dialing via predictive diaelr |
| 01:12.21 | Kobaz | what's your iowait |
| 01:12.39 | Kobaz | high io is a death knell for asterisk |
| 01:13.14 | Oeaa | Cpu(s): 12.1%us, 3.9%sy, 0.1%ni, 83.6%id, 0.1%wa, 0.0%hi, 0.3%si, 0.0%st |
| 01:13.22 | Kobaz | okay, that's good, very good |
| 01:13.25 | Kobaz | it sounds networking-related |
| 01:13.31 | Kobaz | do some mtr |
| 01:13.40 | Kobaz | hit j, check for big swings in jitter |
| 01:13.42 | Oeaa | yeah i'm pretty sure it has to be agent leg, i mean who knows what kinda wifi |
| 01:13.49 | Oeaa | congestion and all that jazz |
| 01:14.25 | Kobaz | Samot: what i'm seeing. is that really... i'll need to reproduce the problem with asterisk core on debug 3 |
| 01:14.37 | Kobaz | pretty much all the sip stuff i would need is using 3 |
| 01:17.44 | Kobaz | or... even better... is strip out all the sip setup into its own debug.. because i don't care about all the other stuff just yet |
| 01:21.20 | Kobaz | oooooh |
| 01:21.27 | Kobaz | looks like that's already done, yay :) |
| 01:21.39 | Kobaz | yay for new-er asterisk.... core set debug 3 chan_sip.c |
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| 13:08.19 | *** join/#asterisk wasanzy (~wasanzy@41.242.137.170) |
| 13:08.25 | wasanzy | hello |
| 13:08.48 | wasanzy | can I assign more than one value to a key in AstDB? |
| 13:10.38 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:10.42 | [TK]D-Fender | wasanzy, no |
| 13:12.33 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:d8e1:ae3d:6fbf:3b50) |
| 13:12.41 | wasanzy | ok thank you |
| 13:12.54 | wasanzy | I would have to use mysql then |
| 13:19.13 | [TK]D-Fender | we don't know. |
| 13:19.21 | [TK]D-Fender | you haven't described an actual need |
| 13:23.27 | wasanzy | I want to build a solution that will require a lot of dialplan code and I want to minimize that by using backend. Reason: I have come up with five levels of categories. |
| 13:23.45 | wasanzy | If there is a way I can send you my diagram, please let me know so you can advice |
| 13:24.02 | [TK]D-Fender | picpaste.com |
| 13:24.08 | [TK]D-Fender | which you should know already.... |
| 13:24.23 | [TK]D-Fender | or any other picture sharing service |
| 13:24.24 | josecapurro | wasanzy: sounds like AGI and a database backend. |
| 13:24.42 | wasanzy | josecapurro: yes |
| 13:24.47 | [TK]D-Fender | not yet it doesn't |
| 13:24.58 | [TK]D-Fender | that is still a hollow answer |
| 13:25.08 | josecapurro | [TK]D-Fender: OK. Sorry. |
| 13:26.04 | [TK]D-Fender | his first sentences said nothing. "5 levels of categories" doesn't say anything meaningful either about the other dimensions of data, or how many unique values, etc |
| 13:28.44 | wasanzy | http://picpaste.com/IVR_SOLUTION-O6i9sE9C.jpg |
| 13:29.03 | wasanzy | please you will have to zoom to see clearly |
| 13:29.20 | wasanzy | I don't want to write so many dialplan for this |
| 13:32.15 | [TK]D-Fender | I see what looks like IVR levels....but it isn't quite clear exactly how much DATA is in there |
| 13:33.11 | [TK]D-Fender | under 1 > 4 (general) > 3 (After Payment) <------------- what happenes here? |
| 13:33.26 | [TK]D-Fender | I don't see where actual data comes in |
| 13:34.46 | wasanzy | [TK]D-Fender: That is the last level in that case. A clip will be played to the caller with the info he or she is requesting for. |
| 13:35.45 | [TK]D-Fender | What "case"? |
| 13:36.04 | [TK]D-Fender | You aren't saying what is VARIABLE in here. |
| 13:36.39 | [TK]D-Fender | Where does DATA come in? So far you are showing what looks like a menu tree. Nowhere are you clearly saying "I'm looking something up here" |
| 13:39.13 | wasanzy | that is a menu tree as you put it. and all the menu are associated with a clip instructing the user what to do at each stage. when the use gets to 1 > 4 (general) > 3 (After Payment) as you pointed out, the After Payment is a clip behind that menu again and will be played to the user |
| 13:39.45 | [TK]D-Fender | Again, where is a DATA LOOKUP? |
| 13:39.53 | [TK]D-Fender | you're describing a MENU with PROMPTS |
| 13:39.58 | [TK]D-Fender | NOTHING in that is VARIABLE. |
| 13:40.19 | wasanzy | I want to build that menu tree by using db. meaning asterisk will query db to retrieve the next menu base on what the user chose and so on till the end of the call. |
| 13:40.20 | [TK]D-Fender | Where is ***DATA*** in there? |
| 13:40.54 | [TK]D-Fender | Because a series of menus with prompts is just an IVR. |
| 13:42.07 | [TK]D-Fender | press 1 (get a fixed prompt), press 4 (get a fixed prompt), press 4 "2012 Retirement". What happens there? |
| 13:42.08 | wasanzy | [TK]D-Fender: Yes is just an IVR but you realized I will have to write a lot of context for this menu tree. But with backend, I may only need two or three context |
| 13:42.36 | [TK]D-Fender | You want to use a DB to DEFINE those levels? |
| 13:42.45 | wasanzy | Retirement: There is a clip that will be played. |
| 13:43.00 | wasanzy | [TK]D-Fender: Yes I want to use a DB to DEFINE those levels |
| 13:44.10 | wasanzy | so asterisk will be dealing with variables instead of fixed values in the dialplan |
| 13:44.27 | [TK]D-Fender | ThAT would be a serious mess |
| 13:44.42 | [TK]D-Fender | Your idea makes no sense to implement in dialplan at all |
| 13:45.02 | wasanzy | [TK]D-Fender: maybe am not explaining well |
| 13:45.02 | [TK]D-Fender | Forget Astb, even the DIALPLAN CODE for this would be a complicated mess for nothing |
| 13:45.14 | [TK]D-Fender | I think I got what you want |
| 13:45.23 | [TK]D-Fender | IMPLEMENTING it would be a complete waste |
| 13:45.41 | [TK]D-Fender | having to look up what level you are at, knowing that some just leavd to SUB menus, etc. |
| 13:45.48 | [TK]D-Fender | Yeah, this is akinda a stupid mess |
| 13:45.59 | wasanzy | why would it be a mess? |
| 13:46.24 | [TK]D-Fender | You don't have a "drill down" set of engine code to power this which is the dialplan you'd have to write to know you're 2 levels deep and it has sub menus |
| 13:46.35 | [TK]D-Fender | and from what you've described I could have written the entire thing by now. |
| 13:46.54 | [TK]D-Fender | a "normal" |
| 13:46.59 | wasanzy | [TK]D-Fender: I have an engine code |
| 13:47.06 | [TK]D-Fender | What out of this is VARIABLE? |
| 13:47.57 | wasanzy | I will implement it and let you know how I solved it. |
| 13:48.09 | [TK]D-Fender | <wasanzy> [TK]D-Fender: I have an engine code <- you can't have this without a DATA SOURCE. And since you were lokoing for one, this doesn't sound possible |
| 13:48.22 | [TK]D-Fender | Your description is still broken. |
| 13:48.51 | [TK]D-Fender | And your goal hasn't been described in a way where having those choices stored adds any VALUE. |
| 13:48.59 | wasanzy | [TK]D-Fender: normally use mysql, but I was thinking of using something else that is why I asked about Astdb |
| 13:49.11 | [TK]D-Fender | nothing in there sounsd VARIABLE. |
| 13:49.19 | [TK]D-Fender | <[TK]D-Fender> press 1 (get a fixed prompt), press 4 (get a fixed prompt), press 4 "2012 Retirement". What happens there? |
| 13:49.52 | [TK]D-Fender | If I go though that chain and ALWAYS get the same stupid recording, then it isn't VARIABLE. Having to go to a DB for that chain doesn't add VALUE. |
| 13:50.11 | [TK]D-Fender | How is this structure or the result CHANGING to give a reason for being in a DB? |
| 13:50.12 | *** join/#asterisk DiegooRamon (~diego@168.194.161.32) |
| 13:50.29 | *** join/#asterisk _douglasqsantos (~douglasqs@168.194.161.32) |
| 13:51.29 | [TK]D-Fender | And then WHO is going to change those values in there? HOW are they going to do it the first time? Then to update it? |
| 13:51.30 | wasanzy | [TK]D-Fender: The recordings are actually not going to be fixed. the menu will not be fixed. they will change. |
| 13:51.37 | [TK]D-Fender | WHEN? |
| 13:51.41 | [TK]D-Fender | by WHOM? |
| 13:51.42 | [TK]D-Fender | HOW? |
| 13:52.05 | [TK]D-Fender | this determines the type of tools you should be using |
| 13:52.31 | wasanzy | This solution is for other people to opt-in. they will be uniquely defined with account numbers. |
| 13:52.34 | [TK]D-Fender | and how often is this expected to change? |
| 13:52.40 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 13:52.47 | [TK]D-Fender | Where do i see account numbers in there? |
| 13:53.03 | [TK]D-Fender | How do I see this flow doing anything different based on that fact? |
| 13:53.11 | [TK]D-Fender | Your description is useless |
| 13:54.26 | wasanzy | [TK]D-Fender: yea is my fault. I didn't indicated that. This is just a menu structure to show how the menus flow. The actual implementation am yet to do a design for that |
| 13:54.44 | [TK]D-Fender | You have not described the need for DATA at all |
| 13:54.59 | [TK]D-Fender | This has been a complete waste of time |
| 13:55.01 | wasanzy | This is not data model is just a flow |
| 13:55.11 | dym | Hey all! Im having an odd problem with my snom370. I moved a working asterisk installation from one machine to another and reconfigured the phones, now all i get is "403" and nothing shows up in the asterisk verbose log. This is the sip log of a call initiation. Does anyone have an idea? https://pastebin.com/raw/tjyDhmW6 |
| 13:55.13 | [TK]D-Fender | You were asking about a DATABASE |
| 13:55.21 | [TK]D-Fender | Why do I give a shit about the MENU? |
| 13:55.27 | [TK]D-Fender | Why are you wasting out time on that? |
| 13:55.51 | [TK]D-Fender | Where are you MANIPULATING DATA?! |
| 13:55.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:56.58 | wasanzy | [TK]D-Fender: The menus are variables. the values will be stored in database. I will build upon the engine I have to manipulate the data |
| 13:57.12 | asteriskmonkey | he wants realtime? |
| 13:57.36 | wasanzy | thanks for your time, I have learn a lot. I will build upon that and get back. |
| 13:57.45 | wasanzy | asteriskmonkey: realtime |
| 13:58.15 | [TK]D-Fender | asteriskmonkey, He wants a way to store enough data to describe a menu flow including going to sub-menus, or to play a prompt. All of that structure to be defined in a DB of some kind |
| 13:58.58 | [TK]D-Fender | From what little he's actually said |
| 13:59.39 | asteriskmonkey | oh he wants to make a commercial product :P |
| 14:00.09 | [TK]D-Fender | asteriskmonkey, It's not entirely clear. He has very little ability to actually describe anything. |
| 14:00.39 | [TK]D-Fender | It not clear what he will use to FILL that tree data in. |
| 14:00.55 | [TK]D-Fender | And that determines what STORAGE might be best for it. |
| 14:01.14 | [TK]D-Fender | And the actual variable naturee of that data should determine f this entire exercise is jsut stupid |
| 14:01.36 | asteriskmonkey | i see, well in that case he needs atleast a 4million dollar oracle licesnse with odb connector :) |
| 14:01.37 | [TK]D-Fender | You don't build an entire pizzaria when you just want 1 stupid pizza. |
| 14:02.54 | *** join/#asterisk [douglasqsantos] (~douglasqs@168.194.161.32) |
| 14:02.59 | [TK]D-Fender | dym, idea: look at the peer it's matching |
| 14:03.30 | [TK]D-Fender | Found peer 'pgsip' for 'pgsip' from 10.1.15.81:1030 |
| 14:04.17 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 14:05.27 | dym | [TK]D-Fender: That's the phones name, registered as friend in sip.conf. |
| 14:06.02 | dym | [TK]D-Fender: theres no peer defined as pgsip. |
| 14:06.29 | dym | [TK]D-Fender: I have one upstream peer defined, which is my online pbx to initiate external calls. |
| 14:06.35 | [TK]D-Fender | <[TK]D-Fender> Found peer 'pgsip' for 'pgsip' from 10.1.15.81:1030 <------ [pgsip] very clearly is |
| 14:06.50 | [TK]D-Fender | "Found peer" |
| 14:06.56 | [TK]D-Fender | you have an entry |
| 14:07.08 | [TK]D-Fender | go look at its definition and run-time status |
| 14:07.33 | dym | does defining something in brackets already count as peer, regardless of the type set? |
| 14:07.48 | [TK]D-Fender | friend = peer + user |
| 14:07.51 | [TK]D-Fender | ^ |
| 14:07.57 | dym | OH |
| 14:09.01 | dym | what is the second "pgsip" in this? Found peer 'pgsip' for 'pgsip', the phone name? i wonder where it clashes. |
| 14:09.01 | [TK]D-Fender | for a call peer only matches on HOST. user would allow to match by the FROM |
| 14:09.14 | [TK]D-Fender | the second is the name the inbound call came in from |
| 14:09.22 | dym | (this worked until 24 hours ago :D, thats why im puzzled) |
| 14:09.27 | [TK]D-Fender | From: "Patrick" <sip:pgsip@10.1.15.5>;tag=uwfwhi4esc |
| 14:09.30 | [TK]D-Fender | THERE |
| 14:09.46 | dym | right, thats the sip username. |
| 14:09.53 | [TK]D-Fender | "used to work" tells us nothing. as I said, LOOK AT IS STATUS NOW |
| 14:10.07 | dym | its not working, thats correct. |
| 14:10.09 | [TK]D-Fender | "sip show peer pgsip" |
| 14:10.13 | [TK]D-Fender | look at the DEFINITION |
| 14:13.22 | dym | Sorry, from my understanding, i had to define a sip user in sip.conf, which then would be classified by type. So using peer instead of user is what seems to be the problem? |
| 14:13.41 | dym | username and the definition are both called pgsip |
| 14:14.47 | dym | https://pastebin.com/raw/cMKD987a |
| 14:14.58 | [TK]D-Fender | I didn't tell you to go make anything |
| 14:15.03 | [TK]D-Fender | I told you to look at what you had |
| 14:15.17 | dym | i didnt change anything :) |
| 14:15.23 | dym | i am looking at what i have. |
| 14:15.32 | dym | i just dont seem to see the problem /: |
| 14:15.50 | [TK]D-Fender | Where's the other thing I told you to look at? |
| 14:16.39 | dym | the general definition of a peer? |
| 14:16.51 | [TK]D-Fender | Also, "username" no longer exists, "canerinvite" should be "directmedia" now, and you shouldn't have an "insecure" on that |
| 14:17.06 | [TK]D-Fender | <[TK]D-Fender> "sip show peer pgsip" <=----------------------------------------- |
| 14:18.18 | [TK]D-Fender | "sip show peers" <- for the full list to see what else you have coming from various hosts |
| 14:18.36 | dym | Yes, im aware of that. |
| 14:18.39 | dym | It's right here: https://pastebin.com/raw/Ecx5HsWj |
| 14:18.44 | dym | I did look at it, just didnt paste. |
| 14:19.36 | [TK]D-Fender | So far looks ok, lets see the full peer dump list |
| 14:20.39 | dym | https://pastebin.com/raw/rJtpf0zD |
| 14:21.21 | [TK]D-Fender | orsip/orsip 10.1.15.81 D Yes Yes 1030 OK (22 ms) |
| 14:21.29 | [TK]D-Fender | pgsip/pgsip 10.1.15.81 D Yes Yes 1030 OK (22 ms) |
| 14:21.33 | [TK]D-Fender | same IP.... |
| 14:21.40 | dym | [TK]D-Fender: Those are 2 sip accounts on the snome phone. |
| 14:21.47 | dym | its deliberate. |
| 14:21.47 | [TK]D-Fender | now we've got some potential for screwups |
| 14:21.52 | [TK]D-Fender | need to look at then both |
| 14:22.01 | dym | sip show peer wise? |
| 14:22.19 | [TK]D-Fender | remove the INSECURE first and fix up the parameters as I've described for bot. retest, and then if still failing, show new debug for both |
| 14:22.44 | asteriskmonkey | anyone got a patch for playback to autotranscode from url targets? |
| 14:23.39 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 14:24.43 | dym | [TK]D-Fender: on it. |
| 14:29.43 | dym | [TK]D-Fender: https://pastebin.com/raw/ffgAnPK6 - the phone has stopped registering altogether. |
| 14:29.43 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:d8e1:ae3d:6fbf:3b50) |
| 14:29.58 | dym | although the peers show up |
| 14:30.10 | [TK]D-Fender | <PROTECTED> |
| 14:30.22 | [TK]D-Fender | looks like a networking or device issue |
| 14:30.33 | [TK]D-Fender | <PROTECTED> |
| 14:30.39 | dym | oh sorry |
| 14:30.40 | dym | here they come |
| 14:30.42 | [TK]D-Fender | because it has an IP on file and is trying to qualify |
| 14:31.09 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 14:31.58 | dym | [TK]D-Fender: Debug. https://pastebin.com/raw/GJK3pt5q |
| 14:43.23 | *** join/#asterisk klictel (~Claude@modemcable009.207-176-173.mc.videotron.ca) |
| 14:44.09 | dym | [TK]D-Fender: Also another paste of peerdetails when available: https://pastebin.com/raw/Jh5JkVkw |
| 14:48.42 | Samot | How come all the devices show connected on LAN/Private IPs but show they are behind NAT and looks like their ports are from router WAN NAT? |
| 14:49.21 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 14:49.57 | Samot | Addr->IP : 10.1.15.81:1032 <--- 103x common WAN NAT port. |
| 14:50.40 | dym | Samot: they are all on the same lan. |
| 14:50.59 | Samot | As Asterisk? |
| 14:51.03 | dym | affirmative. |
| 14:51.13 | Samot | Then why are they showing behind NAT? |
| 14:51.13 | dym | ip addresses of the phones are DHCP |
| 14:51.32 | Samot | Because, they're not. |
| 14:51.58 | dym | sorry? |
| 14:52.23 | dym | they are showing they are behind nat because they are not? |
| 14:52.23 | Samot | nat=force_comedia,rport |
| 14:52.27 | Samot | You have that |
| 14:52.42 | Samot | orsoft/orsoft 10.1.15.74 D Yes Yes 54240 OK (1 ms) |
| 14:52.45 | Samot | ^^^ That |
| 14:52.56 | Samot | Those Yes Yes should be No No if you are on the same LAN |
| 14:53.20 | Samot | Those two settings are NAT settings for the endpoint |
| 14:53.33 | Samot | You're telling Asterisk all your phones are behind NAT |
| 14:53.36 | Samot | Which they are not. |
| 14:53.47 | dym | Samot: the phones and the LAN asterisk server are on the same private network. however the asterisk server itself connects to an upstream asterisk to dialout. |
| 14:53.58 | Samot | OK |
| 14:54.04 | Samot | Still doesn't change what I just said. |
| 14:54.11 | Samot | You have a phone on the same network as ASterisk |
| 14:54.17 | Samot | You've told Asterisk the phone is behind NAT |
| 14:54.20 | Samot | But it's not. |
| 14:54.22 | dym | sip.conf general: https://pastebin.com/raw/QFPv00BM |
| 14:54.47 | dym | nat=yes is incorrect here too? |
| 14:55.00 | Samot | force_rport='No' <-- that's not right |
| 14:55.21 | dym | nat=force_comedia,rport <-- where did you see this setting? |
| 14:55.28 | dym | or did you derive? |
| 14:55.37 | dym | from one of the peers i assume |
| 14:55.54 | Samot | [general] |
| 14:55.54 | Samot | context=incoming |
| 14:55.54 | Samot | force_rport='No' |
| 14:55.54 | Samot | nat=yes |
| 14:56.08 | Samot | force_rport is not a standalone setting |
| 14:56.13 | dym | because the asterisk server itself is indeed behind a nat. |
| 14:56.21 | Samot | Nor should there ever be single quotes around a setting value. |
| 14:56.23 | dym | facing the internet |
| 14:56.29 | Samot | K |
| 14:56.30 | Samot | Again |
| 14:56.32 | Samot | force_rport is not a standalone setting |
| 14:56.33 | dym | okay |
| 14:56.37 | Kobaz | what's the cheapest single port or dual-port fxo that you guys know of |
| 14:56.39 | dym | i've removed that. |
| 14:56.40 | Samot | Nor should there ever be single quotes around a setting value. |
| 14:56.45 | dym | understood. |
| 14:57.03 | Samot | Also |
| 14:57.17 | Samot | nat=yes is the same as nat=force_comedia,rport |
| 14:57.35 | dym | okay |
| 14:57.49 | Samot | What is your local net settings? |
| 14:58.05 | dym | i have the 10.1.15.0/24 only. |
| 14:58.18 | dym | 10.1.15.1 beeing the firewall. |
| 14:58.27 | dym | 10.1.15.5 * |
| 14:59.03 | Samot | OK |
| 15:01.09 | *** join/#asterisk daemonwrangler (~daemonwra@15.179.154.104.bc.googleusercontent.com) |
| 15:07.16 | *** join/#asterisk SpSpBy (~SpSpBy@190.111.231.108) |
| 15:08.53 | *** join/#asterisk daemonwrangler (~daemonwra@15.179.154.104.bc.googleusercontent.com) |
| 15:10.14 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 15:10.14 | *** mode/#asterisk [+o bford] by ChanServ |
| 15:13.20 | dym | Samot: okay - still it doesnt work. any idea on how to proceed? |
| 15:13.32 | *** join/#asterisk kharwell (kharwell@nat/digium/x-jahruiqogtajffmx) |
| 15:13.33 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:21.23 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 15:23.52 | dym | Samot: sorry, in fact it is working now. |
| 15:23.53 | dym | thanks. |
| 15:24.20 | *** join/#asterisk imcdona (~imcdona@2607:f0d8:20:1001:30bd:d471:2b1e:a682) |
| 15:30.17 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:d8e1:ae3d:6fbf:3b50) |
| 15:38.04 | Kobaz | https://www.amazon.com/Obihai-OBi110-Service-Telephone-Adapter/dp/B0045RMEPI/ref=wl_mb_wl_huc_mrai_3_dp?ie=UTF8&pd_rd_i=B0045RMEPI&pd_rd_r=31CS1GXQ2N4EJB035KMJ&pd_rd_w=fZyfM&pd_rd_wg=rLzMK |
| 15:38.08 | Kobaz | that looks a little too cheap |
| 15:43.56 | [TK]D-Fender | "Use with SIP Service ONLY - Does NOT Support Google Voice " <- yup, there goes its value..... |
| 15:44.04 | Kobaz | hehe |
| 15:44.06 | [TK]D-Fender | that was to original point of ever buying one |
| 15:44.16 | Kobaz | for google voice? |
| 15:44.20 | [TK]D-Fender | yup |
| 15:44.56 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 15:45.41 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-qlozsicrvwooejfk) |
| 15:49.03 | *** join/#asterisk dnit (71c11f6e@gateway/web/freenode/ip.113.193.31.110) |
| 15:52.29 | dnit | Hi, I am unable to play music on hold |
| 15:52.30 | dnit | https://pastebin.com/U8fwLd7A |
| 15:53.43 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 16:02.20 | [TK]D-Fender | Go test it directly |
| 16:08.22 | dnit | I tested but its considering on default |
| 16:08.31 | dnit | So when I changed it in default it worked. |
| 16:08.56 | dnit | I meant put the contents of new_moh in default |
| 16:09.34 | [TK]D-Fender | well those contentx look nothing alike so you're clearly getting something wrong in there |
| 16:10.12 | dnit | I wanted to follow https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_BridgeWait |
| 16:10.28 | dnit | So I wrote BridgeWait(,participant,m(new_moh)) |
| 16:11.03 | [TK]D-Fender | go prove your classes are loaded correctly |
| 16:11.30 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 16:12.43 | dnit | Yes they are loaded correctly |
| 16:14.24 | [TK]D-Fender | Time to look at parallel testing of them |
| 16:33.29 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 16:42.55 | *** join/#asterisk Kaian (~kaian@212.81.221.228) |
| 17:16.51 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 17:23.10 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
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| 18:15.07 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
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| 19:11.10 | *** join/#asterisk nny (ad5dbe23@gateway/web/cgi-irc/kiwiirc.com/ip.173.93.190.35) |
| 19:13.09 | nny | Good afternoon all. I am assisting someone with troubleshooting a 3rd party app's inability to auth on the SRTP stream. The error is SRTP unprotect failed with 10/110. I have successfully registered and tested a commerical app (Zopier) and it works fine using the same certificate etc. I do not believe this to be an issue with Asterisk but rather th |
| 19:13.09 | nny | e third party app. The app IS registering with TLS at least. I don't expect much support on this as the nature of the issue is outside asterisk BUT if anyone could tell me specificaly why that error is thrown I can pass it to them and hopefully they can figure it out |
| 19:17.50 | josecapurro | nny: That error is thrown in the Asterisk log/CLI, or in the 3rd party app? |
| 19:18.01 | nny | josecapurro: Asterisk CLI |
| 19:22.44 | josecapurro | nny: Which version of Asterisk? 13? 11? |
| 19:24.02 | josecapurro | nny: It can be problem with the app. I can't find anything on it but some dude saying it was resolved with Asterisk 13. |
| 19:24.22 | nny | josecapurro: Asterisk 13 |
| 19:24.22 | file | it's a libsrtp message, 110 if I recall means that decryption failed based on the key we know |
| 19:25.24 | josecapurro | file: I've found it in res/res_srtp.c, L:415 |
| 19:25.29 | nny | that's how I understand it too. I believe the app may just be messing up the key process |
| 19:26.48 | nny | Zoiper seems to request the key from the server vs. the app which requires the .pem file be uploaded to it |
| 19:26.48 | nny | but the app is succeeding on TLS auth (prior it did not) so it at least handles that key process correctly |
| 19:26.58 | file | TLS and SRTP are unrelated |
| 19:27.19 | file | the SRTP key is passed in the SDP, can be exchanged over any SIP transport but logically makes sense to do it over TLS so it can't be intercepted |
| 19:27.42 | jvwjgames_ | Hey guys again |
| 19:27.56 | jvwjgames_ | outbound and inbound calls are failing |
| 19:28.36 | jvwjgames_ | but because i have TLS active |
| 19:28.53 | jvwjgames_ | all the cpnsole is giving me is Setting global variable 'SIPDOMAIN' to '162.220.209.36 |
| 19:32.09 | *** join/#asterisk nny (ad5dbe23@gateway/web/cgi-irc/kiwiirc.com/ip.173.93.190.35) |
| 19:32.14 | nny | sorry got disconnected |
| 19:32.24 | nny | if anyone responded after my last statement let me know |
| 19:34.04 | jvwjgames_ | can anyone help me |
| 19:34.28 | jvwjgames_ | nny can you help me with TLS |
| 19:34.47 | nny | jvwjgames_: I am working on an issue with it, what is your question? |
| 19:35.05 | jvwjgames_ | i have it setup but when i try to use it it says network busy and the consle spits out Setting global variable 'SIPDOMAIN' to '162.220.209.36 |
| 19:35.40 | nny | that's not an error |
| 19:36.00 | nny | jvwjgames_: is this Freepbx? |
| 19:36.05 | jvwjgames_ | yes |
| 19:36.08 | nny | !freepbx |
| 19:36.13 | nny | whoops |
| 19:36.23 | nny | well, um yeah you should go to #freepbx for support on it |
| 19:36.28 | jvwjgames_ | ok |
| 19:40.29 | nny | josecapurro: sorry bud did you respond after my last message? |
| 19:40.43 | *** part/#asterisk nny (ad5dbe23@gateway/web/cgi-irc/kiwiirc.com/ip.173.93.190.35) |
| 19:40.55 | *** join/#asterisk nny (ad5dbe23@gateway/web/cgi-irc/kiwiirc.com/ip.173.93.190.35) |
| 19:41.01 | *** join/#asterisk Oeaa (~eric@c-73-41-195-206.hsd1.ca.comcast.net) |
| 19:41.23 | Oeaa | hey guys... been having completely random segfaults out of nowhere during production and i cant seem to pinpoint the cause.... any ideas? |
| 19:41.27 | Oeaa | Nov 19 22:07:13 pbx4 kernel: asterisk[4298]: segfault at 8 ip 00a60427 sp b610dd34 error 4 in libc-2.12.so[9ec000+190000] |
| 19:41.27 | Oeaa | Nov 21 02:13:20 pbx4 kernel: asterisk[27918]: segfault at 8 ip 00a60427 sp b17c5e48 error 4 in libc-2.12.so[9ec000+190000] |
| 19:41.27 | Oeaa | Nov 21 20:24:31 pbx4 kernel: asterisk[29810]: segfault at 8 ip 00a60427 sp b299ac34 error 4 in libc-2.12.so[9ec000+190000] |
| 19:42.42 | Oeaa | or any clue how I can root this out and prevent it... I mean it just started happening... no idea why, no changes made.. ugh |
| 19:44.06 | Oeaa | I have gdb attached at the moment so hopefully another one will happen |
| 19:44.21 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 19:44.42 | Oeaa | perfect thank you very much. |
| 19:56.19 | sibiria | is there any sort of guide to what debug level various output has? |
| 19:56.30 | sibiria | f.e. i'd like to get rid of dsp.c's DTMF spam on the debug channel |
| 19:56.52 | sibiria | (but i'd like to keep other debug info) |
| 20:10.44 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
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| 20:26.04 | kharwell | sibiria: none that I am aware. You can take a look in the source file and search for what level the debug message is being outputted at |
| 20:27.00 | kharwell | for instance in https://github.com/asterisk/asterisk/blob/master/main/dsp.c if you look for "ast_debug" you'll see the majority of debug is at level 5, a few at 1, and a couple at 10. So <5 should reduce the number of messages |
| 20:28.08 | kharwell | you can also enable debug level logging for specific modules. See "core set debug" here: https://wiki.asterisk.org/wiki/display/AST/Basic+Logging+Commands |
| 20:28.43 | kharwell | Unfortunately it seems setting the global debug level will override any specific module settings |
| 20:31.30 | sibiria | kharwell: thanks for the advice |
| 20:32.46 | *** join/#asterisk Yiota (~textual@199.33.115.187) |
| 20:35.01 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:d8e1:ae3d:6fbf:3b50) |
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| 20:42.10 | *** join/#asterisk wonderworld (~ww@ip-88-152-174-32.hsi03.unitymediagroup.de) |
| 20:49.49 | wonderworld | i dial multiple local channels with dial(). inside of these local channels, after some redirection checks, i dial SIP peers. i need to set some customCDR fields for the local channel that succeeds to dial a peer which picks up the call. i have no idea on where to put the customCDR, because i seem to have no possibility after a successful dial, and i can't set them before, because they depend on the peer that picked up the call. |
| 20:49.50 | wonderworld | i need to know which peer picked up the call, to set the correct customCDR. |
| 20:51.07 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 20:53.36 | MrTAP | wonderworld: check the U option to run a gosub on the answering channel and implement your logic there |
| 20:55.10 | dym | Samot: And by the next asterisk restart, things were back to "forbidden". I just dont get it. |
| 20:56.25 | wonderworld | MrTAP: Thanks a lot, will have a look. |
| 20:59.00 | dym | Maybe someone else can make sense of this: I have a sipphone with 2 accounts on it registered to my local asterisk. When i try dialing out, i get "forbidden". Please see full config/log: https://pastebin.com/raw/B7eyQx29 |
| 21:01.29 | Samot | dym: Why does all this look like your device is behind NAT |
| 21:01.46 | Samot | dym: Or is the SNOM actually listening on 1034? |
| 21:01.48 | MrTAP | dym: having your devices use the same IP & port for two different users is your issue |
| 21:02.32 | MrTAP | read the default sip.conf file under "DEVICE CONFIGURATION" for some info about type=friend, type=user, type=peer |
| 21:04.09 | dym | MrTAP: It's one device! It's a SNOM 370 with 2 Accounts on it. |
| 21:04.25 | Samot | dym: Or is the SNOM actually listening on 1034? |
| 21:04.48 | dym | would this be udp? |
| 21:05.01 | Samot | This would be in the SNOM |
| 21:05.01 | dym | i did not configure it to. |
| 21:05.09 | Samot | OK |
| 21:05.09 | dym | its randomly chosen, i assume. |
| 21:05.19 | Samot | So please go into the SNOM and tell me what it is set to |
| 21:05.55 | dym | Im looking for some sort of port definition? |
| 21:06.03 | Samot | Yes. |
| 21:06.11 | Samot | Network identity (port): |
| 21:06.31 | Samot | It will be in the Setup -> Advanced -> SIP/RTP section |
| 21:06.34 | Samot | With the sip timers |
| 21:06.48 | dym | Blank. |
| 21:07.45 | Samot | OK. So set that to 5060 |
| 21:08.09 | dym | Done, Rebooting Phone. |
| 21:08.30 | MrTAP | dym: if you do a dialplan show 01728222720@pgsip you get output that makes sense? |
| 21:09.24 | dym | MrTAP: Oddly enough, no. tel*CLI> dialplan show 01728222720@pgsip |
| 21:09.24 | dym | There is no existence of 'pgsip' context |
| 21:09.24 | dym | Command 'dialplan show 01728222720@pgsip' failed. |
| 21:09.31 | MrTAP | Well.... there's your problem |
| 21:09.33 | dym | The dialplan AND the section are there though |
| 21:09.43 | MrTAP | that's where the 403 is coming from |
| 21:09.45 | dym | MrTAP: This is a migrated asterisk setup |
| 21:09.57 | dym | extensions.conf is fully populated. |
| 21:10.08 | *** join/#asterisk sekil (~sekil@cable-89-216-195-64.dynamic.sbb.rs) |
| 21:10.12 | dym | tel*CLI> dialplan show |
| 21:10.12 | dym | -= 0 extensions (0 priorities) in 0 contexts. =- |
| 21:10.14 | dym | wtf |
| 21:10.32 | Samot | STOP |
| 21:10.34 | Samot | FFS |
| 21:10.38 | Samot | That is not the problem |
| 21:10.44 | [TK]D-Fender | Dialplan has nothing to do with this |
| 21:10.58 | Samot | You are trying to find a SIP context in a dialplan context. |
| 21:11.06 | Samot | That is never going to work. |
| 21:11.07 | MrTAP | Samot: look under the sip config |
| 21:11.09 | MrTAP | context=pgsip |
| 21:11.17 | Samot | That's for INCOMING |
| 21:11.24 | Samot | Please follow along. |
| 21:11.38 | [TK]D-Fender | Caller is getting refused |
| 21:11.39 | dym | one second! |
| 21:11.45 | Samot | Also nothing to do with auth |
| 21:11.48 | dym | this could be a simple permissions issue. |
| 21:12.04 | MrTAP | auth is good. INVITE -> 401 -> ACK -> INVITE -> 403 |
| 21:12.09 | Samot | No. |
| 21:12.12 | Samot | 403 = bad |
| 21:12.13 | [TK]D-Fender | 403 = total fail |
| 21:12.17 | Samot | Forbidden. |
| 21:12.19 | Samot | Not allowed |
| 21:12.25 | MrTAP | have ya'll tried placing an outbound call when the sip peer context doesn't exist? |
| 21:12.28 | [TK]D-Fender | 403 = Get the fuck out. |
| 21:12.34 | dym | okay, it's not. |
| 21:12.35 | Samot | INVITE -> 401 -> ACK -> INVITE -> 200 OK |
| 21:12.41 | Samot | ^^ that's how it should be |
| 21:12.48 | Samot | Well with some 1xx's in there too |
| 21:12.54 | Samot | But 403 is a total failure. |
| 21:12.54 | [TK]D-Fender | <MrTAP> have ya'll tried placing an outbound call when the sip peer context doesn't exist? <- caller is REJECTED. Not their requested number. THEM |
| 21:12.57 | sekil | evenin' |
| 21:13.11 | dym | Phone is back, accounts unregistered. |
| 21:13.29 | [TK]D-Fender | Found peer 'pgsip' for 'pgsip' from 10.1.15.81:1034 <- caller is matched against a peer |
| 21:13.32 | sekil | got rid of yeastar finally... |
| 21:13.43 | dym | here we go |
| 21:13.45 | [TK]D-Fender | SIP/2.0 403 Forbidden <- but auth is REJECTED |
| 21:14.01 | dym | I seriously dont know why and how |
| 21:14.03 | dym | but it works now. |
| 21:14.10 | Samot | Right |
| 21:14.12 | Samot | Because.. |
| 21:14.13 | Samot | Ports |
| 21:14.19 | Samot | The same thing I asked hours ago |
| 21:14.28 | Samot | Why did your endpoint look like a random NAT port. |
| 21:14.38 | dym | Do they have to be statically defined? |
| 21:14.41 | dym | Didnt they they did |
| 21:15.04 | Samot | This is an SNOM issue. |
| 21:15.07 | Samot | Really. |
| 21:15.15 | Samot | So yeah, it can't be blank. |
| 21:15.20 | dym | Ah! |
| 21:15.22 | dym | right |
| 21:15.29 | dym | mucho annoying |
| 21:15.39 | dym | Thanks everyone involved/enraged _D |
| 21:15.40 | dym | :D |
| 21:16.16 | Samot | Some phones do not allow unique SIP ports per SIP account. |
| 21:16.20 | Samot | SNOM is one of them. |
| 21:16.25 | Samot | They deal with it at their level. |
| 21:16.51 | dym | I deliberately have a local asterisk server to circumvent any nat issues |
| 21:17.03 | Samot | Plus you are matching on port |
| 21:17.04 | dym | so i have a 1:1 connection from local * to upstream * |
| 21:17.09 | Samot | Plus you are matching on port |
| 21:17.14 | Samot | So that's important. |
| 21:17.16 | dym | on both users? |
| 21:17.20 | dym | did i? |
| 21:17.26 | Samot | The 403 came from ASterisk. |
| 21:17.31 | dym | yeah |
| 21:17.37 | dym | what do you mean by matching on port? |
| 21:17.41 | Samot | Authorization: Digest username="pgsip",realm="asterisk",nonce="3471bd6c",uri="sip:01728222720@10.1.15.5;user=phone",response="7e7fbb23eb11638ed8428bfb08b99101",algorithm=MD5 |
| 21:17.51 | Samot | ^^ There's no port in the URI |
| 21:18.03 | dym | okay |
| 21:25.18 | [TK]D-Fender | packs up to head home... |
| 21:28.15 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 21:35.23 | sekil | panasonic does this too |
| 21:38.21 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 21:58.27 | *** join/#asterisk sekil (~sekil@cable-89-216-195-64.dynamic.sbb.rs) |
| 22:14.10 | *** join/#asterisk Korolev (~Korolev@c-73-179-85-246.hsd1.fl.comcast.net) |
| 22:14.59 | Korolev | how do I get rid of this message: "Correct auth, but based on stale nonce received from peername" |
| 22:15.12 | Korolev | all I've found on google is pedantic=no, but it doesn't change anything |
| 22:20.40 | *** join/#asterisk sekil (~sekil@cable-89-216-195-64.dynamic.sbb.rs) |
| 22:28.27 | MrTAP | Korolev: turn off sip debug and it should go away. it's just a harmless warning |
| 22:29.13 | Korolev | MrTAP, sometimes the device loses registration and it won't register again |
| 22:30.14 | Korolev | they are in the same network segment, I thought that stale nonce was the reason asterisk would refuse registration |
| 22:30.56 | MrTAP | When I have sip debug on I see that message constantly while everything is happily working. So I mean if the device isn't using the new nonce at all, then yeah your device isn't going to work |
| 22:32.13 | Korolev | ok, so you vote for buggy device? |
| 22:33.18 | MrTAP | Without any more info I can't say... sip debugs would help but I don't have the time to help much more |
| 22:33.27 | MrTAP | Just thought I'd let you know that message on it's own doesn't mean there is a problem |
| 22:34.24 | Korolev | thanks! |
| 22:47.45 | Samot | 5:14:59 PM K<Korolev> how do I get rid of this message: "Correct auth, but based on stale nonce received from peername" <-- There's nothing wrong with this message. |
| 22:47.53 | Samot | That means a Re-REGISTER has happened. |
| 22:49.18 | Samot | Korolev: Devices do not "lose" registration really. |
| 22:49.47 | Samot | If Asterisk is dumping the registration of a device that could likely be due to the device not responding to Asterisk's keepalives. |
| 22:50.02 | Samot | The device will not register again until it's expire time is reached. |
| 22:50.08 | Samot | Or you force one. |
| 22:51.23 | Korolev | Samot, thanks, sounds like I need to recheck the network |
| 22:51.29 | *** join/#asterisk sekil (~sekil@cable-89-216-195-64.dynamic.sbb.rs) |
| 22:53.13 | Samot | I'd look to see if that endpoint is going UNREACHABLE |
| 22:53.19 | Samot | That will show up in sip show peers |
| 22:53.35 | Samot | Or even in the console logs |
| 22:54.03 | Samot | Asterisk will mark the endpoint UNREACHABLE and still try to reach it before completely dropping it. |
| 22:54.12 | Samot | But in that state, incoming calls will not work |
| 22:54.15 | Samot | Outgoing will. |
| 22:54.22 | Samot | From the endpoint. |
| 23:05.01 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 23:16.39 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:40.35 | *** join/#asterisk sekil (~sekil@cable-89-216-195-64.dynamic.sbb.rs) |
| 23:56.34 | *** join/#asterisk dar123 (~dar@2602:306:bcbf:e750:d8e1:ae3d:6fbf:3b50) |
