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| 02:46.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 10:27.21 | Tuju | any glue how to make asterisk release log file inodes after logger reload ? |
| 10:27.39 | Tuju | my filesystem is always full until i restart whole service |
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| 11:19.52 | IsUp | hello |
| 11:20.54 | IsUp | i am running 16.0.1 with all of my clients behind firewall and they all use 192.168.1.*, Asterisk is also on local network with the 192.168.1.45 IP which natted to a public IP |
| 11:21.35 | IsUp | using an external provider to make calls over SIP, however i am not sure if i made correct configuration for NAT |
| 11:21.56 | IsUp | i am able to both receive/send calls, no audio issues at all but my calls dropped around 3-5 mins |
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| 12:10.12 | pchero_work | IsUp: Check the why the Asterisk drop the call. There's must be reason. |
| 12:10.48 | pchero_work | I think some firewall or SIP proxy(such as kamailio) has a SIP rule about that. |
| 12:11.14 | pchero_work | Or maybe one of you dialplan sets the TIMEOUT(absolute) settings. |
| 12:11.34 | pchero_work | Check the log first and figure out why. |
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| 12:21.14 | masked | hi |
| 12:21.16 | IsUp | pchero_work: no timeout setting in my dialplan and hangup shows normal with code 16 |
| 12:21.49 | masked | what can i use in place of the Macro app to handle alphanumeric extensions now that the Macro app is deprecated? |
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| 12:24.45 | pchero_work | Is Asterisk hangs up the call first? or other endpoint? |
| 12:27.46 | Samot | masked: GoSub() |
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| 12:38.23 | masked | Samot: nvm i sorted it |
| 12:38.24 | masked | thnks tho |
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| 14:03.42 | csguth | Hi all, |
| 14:03.55 | csguth | Is there a way to accept any registration on Asterisk running as SIP server? |
| 14:04.13 | csguth | Or must I declare all endpoints on pjsip.conf |
| 14:04.13 | csguth | ? |
| 14:05.28 | file | PJSIP does not provide a mechanism to automatically create endpoints/AORs/etc as a result of an incoming REGISTER. The only thing that could be done is to enable the anonymous endpoint, and configure it with an AOR. All unknown endpoints would end up registering to that single AOR. |
| 14:08.34 | csguth | so will the peers be able to call each other? |
| 14:08.55 | file | you wouldn't be able to direct it to a particular peer, no |
| 14:09.59 | file | that use case is counter to what almost everyone using Asterisk wants, and thus noone has recreated the functionality in chan_pjsip to allow it that existed in chan_sip |
| 14:10.19 | csguth | I see |
| 14:11.42 | csguth | ok, so for testing, I'm gonna create the endpoints for each member of my team |
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| 14:12.42 | csguth | I'm currently declaring each one like this: https://pastebin.com/ERJVEgzM |
| 14:12.43 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 14:12.57 | csguth | so I guess I'll have to replicate this for each one |
| 14:13.10 | file | use templates. |
| 14:13.30 | file | https://wiki.asterisk.org/wiki/display/AST/Using+Templates |
| 14:13.36 | csguth | cool |
| 14:13.37 | csguth | perfect |
| 14:13.45 | csguth | I should buy you a beer |
| 14:13.48 | csguth | thanks mate |
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| 14:41.16 | *** mode/#asterisk [+o bford] by ChanServ |
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| 14:51.12 | igcewieling | is it plausible for a destination SIP device to try switching from IPv4 to IPv6 using a SIP 181 Call is Being Forwarded reply sent back to the calling device? |
| 14:59.51 | Samot | Not that I'm aware of. |
| 15:00.29 | file | I've never heard of that being a thing, but it would certainly be a creative way of doing it... |
| 15:00.51 | Samot | I'd be interested in how it does, since 181 replies generally don't have any SDP information |
| 15:01.23 | Samot | 181 is the destination server telling you, we're forwarding the call. |
| 15:01.32 | Samot | It could be due to a CF, CFB, etc. |
| 15:01.34 | file | oh yeah I was thinking of a 302, duh |
| 15:02.00 | Samot | While the server is telling you 181 |
| 15:02.15 | Samot | It's sending the original INVITE out to the new destination |
| 15:03.52 | Samot | igcewieling: Is this actually happening? If so, is it causing an issue? |
| 15:05.46 | igcewieling | Samot: I'm trying to track down the cause of 606 replies. I think I mentioned it the other day. |
| 15:06.22 | igcewieling | since I disabled ipv6 I've not seen another 606, but they are intermittent at best. |
| 15:07.09 | Samot | Did you see if VZ could use a different route for that testing? |
| 15:10.33 | Samot | And do the 606 replies have the additional headers for the reason? |
| 15:12.12 | Samot | They should/might contain a Warning: header that should have a 3XX code in it. |
| 15:12.24 | Samot | And perhaps some text with the reason why it was rejected. |
| 15:12.56 | Samot | Well could be multiple Warning: headers if there are multiple reasons. |
| 15:13.18 | igcewieling | I don't see a reason. screenshot showing headers: https://i.imgur.com/wYvOjQV.png |
| 15:13.56 | igcewieling | this one doesn't have a 181 forward packet and is from before I disabled ipv6 |
| 15:16.23 | *** join/#asterisk Frypolydesoxy (18e201e8@gateway/web/cgi-irc/kiwiirc.com/ip.24.226.1.232) |
| 15:16.33 | Samot | And VZ is saying they are not replying with the 606? |
| 15:16.41 | Samot | They are just "passing it back to you"? |
| 15:17.13 | igcewieling | I see you have sent in trouble tickets to VZB before. 8-| |
| 15:17.32 | Samot | I've dealt with VZ before. |
| 15:17.42 | Samot | Just not in your capacity. |
| 15:18.56 | igcewieling | I deal with Verizon Business (SIP) rather than Verizon (landlines) or Verizon Wireless (cell). |
| 15:19.21 | igcewieling | anyway, I'll leave IPv6 disabled and see if we get any more 606 replies. |
| 15:19.24 | Samot | Who is truly sending the 606? |
| 15:19.36 | igcewieling | the screenshot shows all the info I know about that. |
| 15:20.01 | Samot | So VZB is the .245 address? |
| 15:20.15 | Samot | Sorry, th .18 |
| 15:20.46 | igcewieling | that is a terrible example call. I'll find one with the error going in the right direction. |
| 15:21.02 | Samot | Which is VZ? |
| 15:21.04 | Samot | .245? |
| 15:21.18 | igcewieling | we are .18 |
| 15:21.21 | Samot | Since that is what is sourcing the 606 |
| 15:21.24 | Samot | Who is .245? |
| 15:22.02 | igcewieling | .25 is OrgName: MCI Communications Services, Inc. d/b/a Verizon Business |
| 15:22.13 | igcewieling | .245 I mean. |
| 15:22.28 | Samot | So as far as I can see from this, VZB is replying with the 606. |
| 15:22.46 | igcewieling | 63.64.0.0/10 is the entire block |
| 15:22.53 | Samot | Now are they the actual source of the reply or did they just send it through to you? |
| 15:22.59 | igcewieling | exactly. |
| 15:23.07 | Samot | I'm trying to recap. |
| 15:23.18 | Samot | Because I believe you stated before VZB was claiming it wasn't from them. |
| 15:23.26 | Samot | And it was from the "destination" provider. |
| 15:23.30 | igcewieling | correct. |
| 15:23.49 | Samot | OK, so I said then "See if VZ can use a different route for this destination" |
| 15:23.52 | Samot | Did that happen? |
| 15:24.22 | Samot | I also said to double check if VZ is actually handling the media all the way through... |
| 15:24.25 | igcewieling | After getting that reply on a couple of tickets and because we don't get a lot of these errors, it has not been a high priority. I find it very annoying 8-| |
| 15:24.54 | igcewieling | No. We've not asked VZB to change the routing yet. |
| 15:27.52 | Samot | I take it you weren't sending calls over IPv6 based on these screencaps. |
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| 15:29.04 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:29.18 | igcewieling | the ipv6 idea was only a idea, that is why I asked if it was plausible. |
| 15:29.49 | Samot | Well here's the thing... |
| 15:31.37 | Samot | You peer with VZB. So how you send messages should be confirmed to how VZB wants them. If they are fully handling the transaction, then they are sending you the 606. If they are not and they are proxying/passing through things such as SDP and other information to hop or the actual destination, _they_ may not like it. |
| 15:32.28 | Samot | It would be weird that all the calls work except for X amount of DIDs that all take the same route/destination. |
| 15:32.58 | Samot | So that would mean something is happening along that route/destination and unless you know exactly what, you'll be stabbing in the dark. |
| 15:38.04 | Samot | Here's the kicker too, if the 606 reply is coming from beyond VZB it could possible have the Warning: header(s) in that message and by the time VZB is done with it and "passes" it to you, those headers could be stripped. |
| 15:38.30 | Samot | s/possible/possibly |
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| 18:43.57 | csguth | hey, what characters can I use in the field "username" of pjsip auth? |
| 18:44.07 | csguth | I'm not able to use . |
| 18:44.24 | csguth | like jhon.doe |
| 18:54.41 | igcewieling | - and _ should be OK. |
| 18:55.52 | igcewieling | I know I've used - in endpoint names, I thought I used _ too. |
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| 18:56.44 | csguth | cool |
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| 23:30.39 | degenerate | where can i find documentation about event's for the AMI, for example on hangup event i see things like Cause='16' or ChannelState='6' where can i find the docs that explain what these numbers mean |
| 23:39.36 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:44.39 | Samot | https://wiki.asterisk.org/wiki/display/AST/AGICommand_channel+status |
| 23:44.53 | Samot | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
| 23:50.37 | degenerate | Samot thank you very much |
