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| 00:18.26 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); 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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| 12:06.07 | Guest1031 | Hello everybody! Question: When I perform an outbound call(from an asterisk), I can see that there is no Session header present. However, on the 183 response I see that the upstream asterisk contains a session expires on 1800s and refresher=uas. However, after 15 minutes the upstream asterisk sends an INVITE with a new expires 1800s but refresher is no longer uas but uac. |
| 12:06.29 | Guest1031 | Why would the upstream asterisk change the refresher? Because that INVITE is never forwarded to the softphone and the call just ends after 30 minutes. |
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| 12:09.32 | Guest1031 | Flow: softphone <-> Asterisk <---> Upstream asterisk |
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| 12:51.56 | Samot | There is not "Forwarding" of the INVITE |
| 12:52.39 | Samot | Call from PSTN ->> Upstream Asterisk - Upstream Asterisk DIAL()'s the peer for the main Asterisk. |
| 12:53.02 | Samot | Main Asterisk gets call -- Main Asterisk DIAL()'s the softphone peer |
| 12:53.19 | Samot | So in which part of that flow are you having issues? |
| 12:54.26 | Samot | Also, PSTN to Upstream Asterisk = Channel 1 and Upstream Asterisk to Main Asterisk = Channel 2 |
| 12:54.56 | Samot | This is a B2BUA not a SIP proxy/router/switch |
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| 13:32.26 | Ast001 | Hi, I tried to watch with sip debug why sip device get unregistered by Asterisk 13.31, but when I turn on sip debug that line which say Unregister does not show up for all day like all is ok, but when I turn off sip debug I see those lines again so I can't watch in sip debug for problem. |
| 13:32.42 | Ast001 | 13.21 I meant (newest) |
| 13:33.02 | Samot | The SIP Debug is going to show the SIP Messages. |
| 13:33.05 | Samot | And transactions. |
| 13:33.09 | Samot | So you'll see the REGISTER |
| 13:33.24 | Samot | What you are referring to is output in the verbose log. |
| 13:33.27 | Samot | Two different things. |
| 13:34.39 | Ast001 | I hoped I will see Unregister with sip set debug on and then see above sip message to find reason |
| 13:35.01 | Ast001 | Only interesting thing in debug I saw is NOTICE[857]: chan_sip.c:17309 check_auth: Correct auth, but based on stale nonce received from |
| 13:35.58 | Ast001 | and bellow sip package Unatuhorized and then Register again. |
| 13:37.17 | Ast001 | https://pastecode.xyz/view/37bbe736 |
| 13:38.18 | Ast001 | Can it be that that notice make device unregistered ? |
| 13:39.07 | Samot | No. |
| 13:39.17 | Samot | You have your devices set to register every 60 seconds |
| 13:39.42 | Samot | So they are sending a REGISTER and the system is going "Oh we have a registration for you, let me update this" |
| 13:46.00 | Ast001 | Device has "Re-register Period (s)" and it is 60 |
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| 13:48.44 | [TK]D-Fender | Ast001, Which is what it is doing there. |
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| 14:04.41 | Ast001 | Thing which concern me is that this unregister period last for around 10 seconds during which device is not reachable until registers again. I would like device to be registered all the time. Can I setup some bigger number there (on device) and do I need to change something in sip.conf to make this happen less often ? |
| 14:05.55 | Samot | Asterisk will de-register a peer when it does not respond to qualifies. |
| 14:06.12 | Samot | While all this appears to be on the same LAN |
| 14:06.31 | Samot | I have yet to see anyone try an use a GOIP device without major headaches or it being a real pain in the ass. |
| 14:06.37 | Samot | Because they're pretty cheap. |
| 14:08.32 | [TK]D-Fender | Step 1: stop using "qualify" |
| 14:08.51 | [TK]D-Fender | We also do not see anything counting it as "unregistered" |
| 14:08.57 | Ast001 | Yes it is in same LAN. No nat between so I used qualify no |
| 14:09.03 | [TK]D-Fender | And if it's there on your LAN you should stop registering PERIOD |
| 14:09.39 | Ast001 | but that all happens with qualify=no |
| 14:10.02 | [TK]D-Fender | <[TK]D-Fender> And if it's there on your LAN you should stop registering PERIOD <----- |
| 14:10.53 | Ast001 | You mean to put host=goipip and to remove username and secret from peer config ? |
| 14:11.29 | [TK]D-Fender | partly |
| 14:11.43 | [TK]D-Fender | who said anything about removing AUTH from calls? |
| 14:11.59 | [TK]D-Fender | Just tell it to stop registering and put the host in directly |
| 14:15.32 | Ast001 | I think goip must send registration because even if i delete username and secret in voip setup it still sends some registration to asterisk. |
| 14:16.10 | [TK]D-Fender | tell it to stop |
| 14:16.28 | Ast001 | Ok I will try to find out how. |
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| 16:22.58 | robl^ | Hrmmm, look, I found my old old old Grandstream Budgetone 101. I wonder if I should deploy to our CEO |
| 16:29.41 | [TK]D-Fender | It can hold down a small stack of paper from a casual breeze or act as a door-stop |
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| 17:21.03 | robl^ | [TK]D-Fender I'd think it make a poor doorstop, unless I injected it with some molten lead to add a bit of weight |
| 17:21.48 | [TK]D-Fender | Well... it'd be limited at best... |
| 17:21.55 | [TK]D-Fender | Stupid BarbieTones... |
| 17:23.38 | igcewieling | Don't put anything important in the old grandstream box. <-- learned the hard way. |
| 17:50.13 | theGoat | i am trying to compile dahdi on a fresh install of ubuntu 18.04 and keep getting these errors: https://pastebin.com/rv9b6vMc. has anyone gotten it to compile on 18.04 yet? |
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| 20:45.01 | DannyA | not an asterisk question, but looking for some SIP help if someone is able to: i got a Polycom Soundstation IP 500, using it at home with my TPLink router, using Twilio SIP. Outgoing calls work just fine, and I'm registered with the SIP domain, however I can only receive incoming calls for the first 5 minutes after the phone boots up, and after that the Twilio server says it can't reach my phone. |
| 20:45.17 | DannyA | There's no settings in Twilio, the phone, or my router for UDP timeout (as far as I can tell) |
| 20:45.41 | DannyA | when I open port 5060 on my router it works perfectly, never drops, however, i constantly get these phantom incoming calls from caller ID 1001 |
| 20:45.43 | DannyA | any ideas? |
| 20:50.29 | igcewieling | DannyA: try setting keepalive=20 in your sip.conf |
| 20:50.45 | DannyA | where would that go? |
| 20:50.50 | igcewieling | that will keep traffic flowing but doesn't generate a lot of system resource use. |
| 20:51.03 | DannyA | i dont have any place to configure files like that (as far as i know) |
| 20:51.07 | igcewieling | /etc/asterisk/sip.conf in [general] (assuming you are not running a GUI) |
| 20:51.08 | DannyA | the phone has a basic web interface |
| 20:51.20 | DannyA | does the phone have a way to access files behind the scenes? |
| 20:51.24 | igcewieling | ah, I see in the scroll back. |
| 20:51.37 | igcewieling | nevermind then. |
| 21:04.05 | Samot | TPLink router <-- That's pretty much the problem. |
| 21:06.01 | Samot | DannyA: You can REGISTER fine but if Twilio is dropping you or not seeing you respond to them, the the TPLink is messing up the NAT. |
| 21:06.21 | Samot | DannyA: When you just open port 5060 UDP to the world, you're opening yourself to get SIP scanned. |
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| 21:22.39 | DannyA | @samot but it does work, just fgor 5-10 mins |
| 21:23.00 | Samot | Right and the the TPLink looses its mind over the NAT |
| 21:23.08 | Samot | This is not uncommon. |
| 21:23.28 | DannyA | damn |
| 21:23.29 | Samot | Make sure SIP ALG is disabled if it is a setting on the TPLink |
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| 21:24.26 | DannyA | https://cl.ly/2s0T0v3m3k3J |
| 21:24.31 | DannyA | screenshot of the settings |
| 21:24.34 | DannyA | it doesnt have sip setting :( |
| 21:25.27 | Samot | Oh dear god |
| 21:25.32 | Samot | Turn off SPI |
| 21:25.49 | DannyA | the firewall setting? |
| 21:25.57 | DannyA | will that help? |
| 21:25.57 | Samot | That's the only option? |
| 21:26.10 | DannyA | there's nothing that says SIP anywhere in these settings |
| 21:28.51 | Samot | https://www.balticnetworks.com/mikrotik-5-gigabit-port-dual-core-880mhz-ethernet-router-w-sfp-level-4.html |
| 21:28.56 | Samot | That's the best I can offer you |
| 21:29.11 | Samot | My general experience with sites that have TPLinks is to replace the TPLinks. |
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| 22:44.38 | DannyA | is there a better router thats more sip friendly? |
| 22:48.31 | jpsharp | We recommend fortigate 80s to our clients. They'll easily handle 90-100 SIP & webrtc users behind them. |
| 22:49.37 | Samot | DannyA: I posted a link. |
| 22:50.21 | Samot | jpsharp: We were just dealing with a gut and a Fortigate earlier today in #freepbx |
| 22:50.27 | Samot | Pretty much the same issues. |
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| 22:56.34 | DannyA | got it, thanks! |
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