| 01:38.39 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 01:49.32 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
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| 03:21.48 | pbp_w | hello, can someone tell me how the DID is detected for sip ? |
| 03:21.59 | pbp_w | or send by the carrier |
| 03:22.14 | *** join/#asterisk defsdoor_ (~Andrew@46.16.211.156) |
| 03:25.01 | *** join/#asterisk ^Gecko^ (asdjhasd@gateway/vpn/privateinternetaccess/gecko/x-95565820) |
| 03:25.26 | ^Gecko^ | I'm having trouble setting up a Cisco 7941G with asterisk. I got the phone sort of working, but it keeps restarting itself during registration. the only errors in the log that show up are error updating locale, and no trust list installed.; load file is SIP41.9-4-2sr3-1s |
| 03:25.38 | ^Gecko^ | phone config: https://pastebin.com/nZcE4DCG |
| 03:50.56 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 04:19.45 | ^Gecko^ | any ideas? |
| 04:44.39 | *** join/#asterisk troyt (zncsrv@2601:681:4100:8981:44dd:acff:fe85:9c8e) |
| 04:50.17 | *** join/#asterisk setham (~textual@unaffiliated/setham) |
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| 06:14.46 | *** join/#asterisk AlfaGulf (5ebbee04@gateway/web/freenode/ip.94.187.238.4) |
| 06:15.24 | AlfaGulf | hello |
| 06:22.42 | Samot | ^Gecko^: That's a question for the Cisco forums more than here. |
| 06:59.11 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 07:11.01 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 07:15.45 | *** join/#asterisk pchero_work (~pchero@87.213.240.121) |
| 07:26.45 | AlfaGulf | <PROTECTED> |
| 07:34.09 | pchero_work | AlfaGulf: What happen? |
| 07:34.37 | pchero_work | Need full logs and version info. |
| 07:38.18 | pchero_work | AlfaGulf: May sngrep from the Asterisk server would be helpful. Check the sngrep about the RTP sending. Does Asterisk receive/send the RTP packets or correct SIP messages? |
| 07:54.53 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 08:03.05 | *** join/#asterisk WHiZZi (~bart@2a00:5140:3741:bb01:6c28:51ff:33c0:ea3b) |
| 08:08.24 | WHiZZi | Goodday all. Quick question, can I do a 'dialplan add' for a switch to Realtime ? |
| 08:12.37 | pchero_work | Not clear, what is the switch? |
| 08:13.59 | *** join/#asterisk mattchis6 (~mattchis@c-107-2-189-89.hsd1.co.comcast.net) |
| 08:14.37 | WHiZZi | something like 'switch => Realtime/context@realtime_ext |
| 08:14.51 | AlfaGulf | The Sip Client is Nexus 6P with google stock Sip Client. The server is behind NAT. |
| 08:15.48 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:16.27 | AlfaGulf | Packet capture shows the Server starts immidiatly sending RTP packets to client, but client start sending RTP packets after exactly 8 seconds from establishing the call. |
| 08:17.19 | *** join/#asterisk elguero (~miguel323@74-95-21-41-Connecticut.hfc.comcastbusiness.net) |
| 08:18.46 | AlfaGulf | During the first 8 seconds, neither side hears the other. |
| 08:25.00 | pchero_work | AlfaGulf: Doesn't look like Asterisk's problem, as long as Asterisk sending a RTP packets. How about check the RTP transmission from the Nexus6 first? |
| 08:25.15 | AlfaGulf | here is the flow sequence: https://drive.google.com/open?id=1MbK3wsCRAIQUewoFzIgCne8PJRlCAvDu |
| 08:25.31 | pchero_work | Then you could know this is some kind of firewall problem between Asterisk and Nexus6 or not. |
| 08:27.52 | *** join/#asterisk devdvd_ (17f248c1@gateway/web/freenode/ip.23.242.72.193) |
| 08:28.10 | AlfaGulf | I am using Mikrotik router, the above capture was done on the router, I can see that the WAN port of my router is passing the RTP packet it gets from the server to the client, but I can't see any reply from the client for the first 8 seconds. |
| 08:28.58 | pchero_work | AlfaGulf: Strange. The problem is Nexus6 doesn't send the RTP. |
| 08:29.05 | AlfaGulf | I tried several packet capture on my nexus 6P but they all failed to capture, can you suggest a reasonable app? |
| 08:29.20 | devdvd_ | Hello, does asterisk have a built in way to monitor the status of multiple channels in a sip trunk. For example, I have a single connection to a trunk provider with 5 channels. Is there a way to monitor the status of each of those channels to see if they are in use? |
| 08:29.42 | pchero_work | AlfaGulf: Zoiper? |
| 08:31.50 | AlfaGulf | Zoiper looks like a SIP Client, does is do packet capture as well? |
| 08:32.53 | AlfaGulf | Sorry guys, have to go now, I'll check later, thanks |
| 08:44.53 | pchero_work | devdvd_: Are you talking about capacity of the trunk? If it is, there's no way to checking the capacity in a easy was as far as I know. |
| 08:45.32 | pchero_work | You have keep focus on the how many available channels are left all the time. |
| 08:46.17 | pchero_work | Check the available channels before creating channel, and change the available counters after creating/terminating channels. |
| 08:54.00 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 09:09.40 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 09:17.01 | *** join/#asterisk Posterdati (~Posterdat@95.238.224.91) |
| 09:17.04 | Posterdati | hi |
| 09:37.26 | *** join/#asterisk sa02irc (~sa02irc@212.43.79.155) |
| 09:53.05 | AlfaGulf | I'm back |
| 09:56.28 | AlfaGulf | Please note that if the SIP Clieint (Nexus 6P) is connected to the same LAN via wifi (ie not NAT), then audio is heard immediately. |
| 10:05.03 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 10:15.57 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 10:18.28 | pchero_work | AlfaGulf: I'm doubting about the SIP client. How about test normal machine like laptop first with sngrep or wireshark tools? |
| 10:20.33 | Posterdati | Samot: hi! |
| 10:23.08 | Posterdati | please help, which are the ports to forward to the PBX (pc running OpenBSD + asterisk)? I forwarded 10000-20000 (udp and tcp) and 5060-5070 (udp and tcp) |
| 10:23.14 | Posterdati | thanks |
| 10:24.59 | pchero_work | Depends on your sip.conf or pjsip.conf configuration. 5060 is default for listen and would be any ports between 10000 ~ 20000 for temp port. |
| 10:26.52 | Posterdati | I forwarded 10000-20000 (tcp and udp) too |
| 10:28.14 | Posterdati | provider <--> 887va (forward 5060-5070 and 10000 20000 to pbx lan ip) <--> pbx |
| 10:28.53 | Posterdati | I checked open ports on pbx runninx nc -l 10000 and using https://canyouseeme.org/ on the same port |
| 10:29.59 | AlfaGulf | Good idea, I'll setup a laptop as a sip client tethered to my mobile and capture to see if client is sending RTP packets on time. |
| 10:48.58 | Posterdati | AlfaGulf: and was you able to see packets? |
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| 12:38.01 | *** mode/#asterisk [+o bford] by ChanServ |
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| 13:51.10 | *** join/#asterisk AlfaGulf (5ebbee04@gateway/web/freenode/ip.94.187.238.4) |
| 13:53.15 | AlfaGulf | pchero_work: I have captured the packets from the client side and found that the client is sending rtp packets immediately but not receiving any rtp from server before 8 seconds. |
| 13:54.14 | AlfaGulf | it looks like my ISP is blocking rtp traffic for the initial 8 seconds |
| 13:55.49 | pchero_work | Congrats. :) Then ask to ISP how to make this possible. |
| 13:55.49 | pchero_work | Or use proxy/vpn whatever. :) |
| 13:56.39 | *** join/#asterisk ashka (~postmaste@pdpc/supporter/active/ashka) |
| 13:56.39 | AlfaGulf | have you seen such case with ISPs? |
| 13:56.48 | Samot | No.\ |
| 13:57.30 | AlfaGulf | I wonder if they are intentionally doing this or they simply un-aware of this |
| 13:57.50 | Samot | Show this traffic... |
| 13:58.10 | Samot | What makes you think that your ISP is blocking random UDP traffic for 8 seconds? |
| 13:58.48 | AlfaGulf | because when i used another isp, I had no problem |
| 13:59.26 | AlfaGulf | Ãا٠ÙشةخÙ٠ةشغ ﻻث غخع Ø´ÙØ« Ùخ٠شصشÙØ« خب ØÙثرÙخعس ؤخÙرثÙسشÙÙخ٠|
| 13:59.29 | Samot | And the same equipment? |
| 13:59.41 | Samot | Oh wait, the other ISP would have a different CPE as well. |
| 13:59.46 | AlfaGulf | sorry, keyboard went into another language. |
| 13:59.49 | Samot | So, again so this traffic. |
| 13:59.57 | Samot | show |
| 13:59.58 | AlfaGulf | yes same equipment |
| 14:00.06 | AlfaGulf | wait |
| 14:03.53 | AlfaGulf | this is an example of traffic seen on the server side (router wan port) : https://drive.google.com/open?id=1T07nZbstyLLV_vLDvw0lyMxry-0TDW4g |
| 14:04.33 | AlfaGulf | This an example of traffic seen on the client side: https://drive.google.com/open?id=1Z7yCegpV9ChHV7E6OsSMaml7pZu4a-Tn |
| 14:05.32 | AlfaGulf | from both sides, rtp traffic is leaving source but not reaching destination only after 8 seconds |
| 14:06.02 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 14:08.22 | pchero_work | I had a same problem before, most of Korea ISP blocked the some of the known-ports basically, but that was pretty long time ago(8 years ago?!). |
| 14:11.01 | Posterdati | [TK]D-Fender: I forwarded 5060-5070, 10000-20000 both tcp and udp to the pbx, I can check that with nc -l 10000 on the pbx and https://canyouseeme.org/ |
| 14:11.36 | [TK]D-Fender | 10000-20000 is ONLY UDP |
| 14:11.57 | Posterdati | ok |
| 14:12.07 | [TK]D-Fender | [TK]D-Fender> <[TK]D-Fender> permit tcp any any range 10000 20000 <- RPT is over UDP, so this is bad |
| 14:12.22 | Posterdati | ah |
| 14:12.24 | [TK]D-Fender | Why does everything have to be repeated 5+ times? |
| 14:13.54 | *** join/#asterisk waldo323_ (~waldo323@75-151-31-89-Michigan.hfc.comcastbusiness.net) |
| 14:13.56 | Posterdati | who knows |
| 14:17.02 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 14:17.43 | Posterdati | [TK]D-Fender: https://paste.debian.net/1072228/ |
| 14:18.34 | [TK]D-Fender | Shows nothing good |
| 14:18.43 | [TK]D-Fender | looks like you've screwed up your firewall |
| 14:18.49 | Posterdati | [TK]D-Fender: as I start asterisk from console |
| 14:19.11 | [TK]D-Fender | taht says nothing about having taken any action |
| 14:19.14 | [TK]D-Fender | DO SOMETHING |
| 14:19.18 | [TK]D-Fender | and check the result |
| 14:19.29 | Posterdati | [TK]D-Fender: ports 5060-5070 and 10000-20000 are forwarded |
| 14:19.43 | Posterdati | to 10.0.0.2 which is the pbx |
| 14:20.04 | [TK]D-Fender | Where's the call? |
| 14:21.18 | Posterdati | https://paste.debian.net/1072229/ |
| 14:21.20 | Posterdati | jere |
| 14:21.22 | Posterdati | here |
| 14:21.26 | *** join/#asterisk adhawkins (~adhawkins@musicbrainz/user/adhawkins) |
| 14:21.56 | Samot | telecomitalia/+3906244052 156.54.82.96 No No 5060 UNREACHABLE |
| 14:22.18 | Samot | Retransmitting #3 (no NAT) to 5.97.53.10:5060: |
| 14:22.20 | Posterdati | Samot: :) yes is what I tried to say... |
| 14:22.30 | AlfaGulf | I just talked to my ISP tech support, they initially denied any blocking, but when I explained to them the capture results, they promised to investigate and let me know. |
| 14:23.09 | Posterdati | Samot: if I remove externaddr=x.x.x.x I can register the telecomitalia peer |
| 14:23.24 | Samot | Posterdati: Do you have a static WAN IP? |
| 14:23.30 | [TK]D-Fender | Posterdati, You've clearly screwed up your firewall. packets aren't getting back from your provider and they were fine before |
| 14:23.41 | [TK]D-Fender | Posterdati, Go learn how to actually maintain that thing |
| 14:23.53 | Posterdati | Samot: no, dynamic ip |
| 14:24.01 | Samot | Uhm.... |
| 14:24.09 | Samot | Then you can't set a static external IP |
| 14:24.13 | Samot | Can you? |
| 14:24.16 | [TK]D-Fender | you can |
| 14:24.25 | [TK]D-Fender | and when it changes you'll be lying to the other side |
| 14:24.29 | [TK]D-Fender | which will be bad |
| 14:24.33 | [TK]D-Fender | but until then will work |
| 14:24.36 | Samot | No kidding. |
| 14:24.44 | Samot | That was the point of my question. |
| 14:24.44 | [TK]D-Fender | enough to fix the rest of the junk |
| 14:24.48 | Samot | But for him to answer. |
| 14:25.15 | [TK]D-Fender | Posterdati, Show your full current config |
| 14:25.26 | Posterdati | I mean if I place externaddr=connection_ip |
| 14:25.31 | Samot | Posterdati: So all of this to connect a single phone to a provider? |
| 14:25.34 | Posterdati | it does not register |
| 14:25.43 | Posterdati | Samot: yes |
| 14:25.45 | wyoung | [TK]D-Fender: You're a champ mate! Keep up the good work! |
| 14:25.54 | Samot | But a new provider or a new phone isn't an answer? |
| 14:26.03 | [TK]D-Fender | wyoung, This one seems pointless... |
| 14:26.07 | Posterdati | Samot: I told you... No... |
| 14:26.09 | Samot | How is that even possible? |
| 14:26.20 | Samot | Did you not check things before buying either? |
| 14:26.28 | wyoung | [TK]D-Fender: perhaps but you are very patient :) |
| 14:26.30 | Posterdati | I cannot change the provider |
| 14:26.36 | Samot | So then change the phone. |
| 14:26.38 | Posterdati | and the phones are brand new |
| 14:26.41 | Samot | So? |
| 14:26.45 | Samot | They are not good. |
| 14:26.47 | wyoung | Posterdati: Fun |
| 14:26.50 | Samot | It doesn't matter that they are new. |
| 14:26.53 | Samot | They are poor phones. |
| 14:26.55 | Samot | Overall. |
| 14:26.57 | [TK]D-Fender | <[TK]D-Fender> Posterdati, Show your full current config |
| 14:27.01 | Posterdati | the phones are ok, is the provider which is not! |
| 14:27.13 | [TK]D-Fender | And prove your WAN IP is the same |
| 14:27.13 | Samot | So you have a poor provider but you can't switch them. |
| 14:27.19 | Posterdati | it uses a 64 characters password |
| 14:27.29 | Posterdati | Samot: yes |
| 14:27.53 | Samot | Well then I suggest you start learning some networking and how Asterisk works properly |
| 14:28.08 | Samot | Because that is going to be an issue. Your router is clearly not setup properly for this. |
| 14:28.52 | [TK]D-Fender | Samot, he was getting comms from them regardless of his "externaddr" before. He's screwed his router config up or something else new |
| 14:28.56 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 14:29.19 | Samot | You've been in here for a week and you've consumed a lot of time here from people. With a lot of that consumption being the same things repeated over and over to you because you will not follow directions. |
| 14:29.20 | Posterdati | https://paste.debian.net/1072230/ |
| 14:29.24 | Samot | So start. |
| 14:29.39 | [TK]D-Fender | externadd=95.238.224.91:5060 |
| 14:29.43 | [TK]D-Fender | Missing an "r" |
| 14:29.45 | AlfaGulf | Thanks every one, I'll update you if/when I get any response from my ISP. |
| 14:29.47 | [TK]D-Fender | and don't put the port there |
| 14:29.50 | [TK]D-Fender | 2 NEW fuckups |
| 14:29.58 | [TK]D-Fender | Stop inventing garbage |
| 14:30.34 | Posterdati | same problem |
| 14:30.44 | Posterdati | externaddr does not work |
| 14:31.15 | [TK]D-Fender | Show me a fixed config and "sip show settings" |
| 14:31.30 | *** join/#asterisk tmoore (~tmoore@50-253-243-17-static.hfc.comcastbusiness.net) |
| 14:31.39 | [TK]D-Fender | <[TK]D-Fender> And prove your WAN IP is the same |
| 14:32.02 | *** join/#asterisk adhawkins (~adhawkins@musicbrainz/user/adhawkins) |
| 14:32.23 | Posterdati | sip show settings --> https://paste.debian.net/1072231/ |
| 14:32.58 | Posterdati | 95.238.224.91 taken from https://canyouseeme.org |
| 14:33.35 | [TK]D-Fender | You shouldn't need an external service to see what IP your router uses |
| 14:34.07 | Posterdati | https://paste.debian.net/1072233/ |
| 14:34.55 | Posterdati | show interface Dialer0 --> Internet address is 95.238.224.91/32 |
| 14:35.09 | [TK]D-Fender | "iptables --list" |
| 14:36.05 | Posterdati | I'm not running the pbx on linux |
| 14:36.16 | Samot | Pardon? |
| 14:36.38 | [TK]D-Fender | Whatever |
| 14:36.43 | Samot | Oh this is OpenBSD..wooo. |
| 14:36.43 | [TK]D-Fender | show the equivalent |
| 14:36.50 | [TK]D-Fender | and then show your actual firewall config |
| 14:37.19 | [TK]D-Fender | What did I tell you about 1 line responses when asked for full dumps & configs? |
| 14:37.54 | Posterdati | Samot: are you a windows 10 fan? |
| 14:38.11 | Samot | I'm a user. Fan is a strong word. |
| 14:38.21 | Posterdati | ah ok |
| 14:38.26 | Posterdati | good luck |
| 14:38.34 | Samot | I don't know what that means. |
| 14:39.36 | Samot | Posterdati: Let me ask you this. Did you ask the provider if you can have your password changed to something the phone supports? |
| 14:39.44 | [TK]D-Fender | Luck is what people who don't know what they're doing rely on. |
| 14:39.54 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 14:40.14 | Posterdati | Samot: they won't change it |
| 14:40.30 | Samot | OK. |
| 14:40.39 | Samot | So you've basically put yourself in a corner. |
| 14:40.45 | Samot | And will do nothing of logic to get out. |
| 14:40.48 | Samot | Got it. |
| 14:41.23 | Posterdati | Samot: this is not the case, anyway |
| 14:41.30 | Samot | It's not? |
| 14:41.38 | Samot | You have a provider that you claim is doing non-standard things |
| 14:41.46 | Samot | You have a phone that is be a POS since you got it. |
| 14:41.57 | Samot | But this, this is the way to go. |
| 14:42.51 | [TK]D-Fender | phone isn't even the issue here. |
| 14:42.58 | [TK]D-Fender | He can't deal with basic networking |
| 14:43.26 | Posterdati | [TK |
| 14:43.33 | [TK]D-Fender | Doesn't matter to me if the phone and provider don't work well together if he can't even manage his own entwork |
| 14:43.44 | *** join/#asterisk m4rcu5 (nobody@546AF885.cm-12-3d.dynamic.ziggo.nl) |
| 14:43.52 | Posterdati | [TK]D-Fender: or you didn't listen to me when I told you my network topology |
| 14:44.32 | [TK]D-Fender | Maybe I was gone when you continued with others. |
| 14:44.39 | [TK]D-Fender | but feel free to re-describe it |
| 14:44.50 | Posterdati | provider <--> cisco 887va <--> pbx + phone |
| 14:44.55 | [TK]D-Fender | And we'll see if I think it should matter... |
| 14:45.30 | [TK]D-Fender | So far... nothing special |
| 14:45.33 | [TK]D-Fender | and? |
| 14:46.33 | Posterdati | and the issue should be the cisco firewall rules since the pbx seems fixed as per your kindly configuration |
| 14:46.34 | [TK]D-Fender | Your provider was talking to your PBX regardless of any of these settings we've been trying to fix in your SIP config |
| 14:46.42 | [TK]D-Fender | which means you've screwed up core netwroking |
| 14:46.50 | [TK]D-Fender | and we're not looking at that right now. |
| 14:46.59 | [TK]D-Fender | Which says there is a serious focus problem |
| 14:48.03 | Posterdati | I do not think that network is screwed as you said, since actually is working, ports forwarded to pbx... |
| 14:48.38 | Samot | So the trunk is no longer UNREACHABLE? |
| 14:48.46 | Samot | You can make and receive calls? |
| 14:48.47 | Posterdati | still unreachable |
| 14:48.52 | Posterdati | Samot: no |
| 14:48.52 | Samot | Then it's not working. |
| 14:48.59 | Samot | Because the packets are not making it back to the PBX |
| 14:49.02 | Posterdati | but networking IS |
| 14:49.04 | Samot | So that is a networking issue. |
| 14:49.14 | Samot | No it is not. Not fully and not properly for SIP traffic. |
| 14:49.16 | Posterdati | but ports are forwarded |
| 14:49.27 | Samot | OK |
| 14:49.38 | Samot | Have you done any troubleshooting at the network level? |
| 14:49.47 | Samot | Are the packets being forwarded properly? |
| 14:50.00 | [TK]D-Fender | Why should we trust that it's correct? |
| 14:50.00 | Samot | Just because you have a forward rule doesn't mean it's working as you think. |
| 14:50.03 | [TK]D-Fender | where's the PROOF? |
| 14:50.16 | Samot | Everything that is an issue right now is a NAT/network issue. |
| 14:50.17 | [TK]D-Fender | How do know all the IP's are right? |
| 14:50.19 | Posterdati | ok, how can I reliably test for them? |
| 14:50.19 | [TK]D-Fender | or ports |
| 14:50.20 | [TK]D-Fender | or anything? |
| 14:50.35 | Posterdati | I asked to you! |
| 14:50.36 | Samot | We can't teach you how to troubleshoot your Cisco 887 |
| 14:50.42 | [TK]D-Fender | If you can't tell what IP your router has then go learn how to manage it |
| 14:51.04 | Posterdati | I forwarded 5060-5070 and 10000-20000, plus I put in rtp.conf rtpstart and rtpend to match 10000-20000 range |
| 14:51.10 | Samot | OK |
| 14:51.19 | Samot | But the router isn't doing something right. |
| 14:51.19 | [TK]D-Fender | you got it wrong last time |
| 14:51.24 | Samot | So you need to check it. |
| 14:51.25 | Posterdati | Samot: could be! |
| 14:51.36 | [TK]D-Fender | and were running off blogs that tell me you don't actually know what you're doing with it |
| 14:51.45 | [TK]D-Fender | "could be"? |
| 14:51.46 | [TK]D-Fender | No |
| 14:51.48 | [TK]D-Fender | IS |
| 14:51.50 | Posterdati | ok, now this is a cisco router, for which I have zero experience in config it |
| 14:52.01 | [TK]D-Fender | You were getting packets before and the you screwed something up |
| 14:52.07 | [TK]D-Fender | GO LEARN |
| 14:52.46 | Posterdati | no I've got packets because I didn't use externaddr in [general]. Stop/ |
| 14:53.13 | Posterdati | if I remove that line, peer register and I can call and recieve |
| 14:53.24 | [TK]D-Fender | Lets see.... |
| 14:53.27 | Posterdati | yes |
| 14:53.46 | [TK]D-Fender | go show us |
| 14:57.14 | Posterdati | icoming call --> https://paste.debian.net/1072236/ |
| 14:57.42 | [TK]D-Fender | ... |
| 14:57.46 | [TK]D-Fender | SIP DEBUG <_ |
| 14:57.49 | [TK]D-Fender | full damn call |
| 14:58.38 | Posterdati | https://paste.debian.net/1072237/ |
| 14:59.30 | Posterdati | this was an incoming call |
| 14:59.49 | Posterdati | now the peer is reachable |
| 15:00.03 | Posterdati | but I cannot place a call from the phone! |
| 15:00.07 | [TK]D-Fender | full damn call |
| 15:00.08 | [TK]D-Fender | ^ |
| 15:00.28 | Posterdati | I hung up from the sip phone |
| 15:00.38 | [TK]D-Fender | full damn call <----------- |
| 15:03.00 | Posterdati | https://paste.debian.net/1072238/ |
| 15:03.11 | Posterdati | this is another incoming call |
| 15:05.29 | [TK]D-Fender | No matching peer for '+393293516963' from '5.97.53.10:5060' |
| 15:05.36 | [TK]D-Fender | The peer you set up for them doesn't match |
| 15:05.52 | [TK]D-Fender | So those settings in it aren't even getting used |
| 15:05.57 | [TK]D-Fender | just for starters |
| 15:08.06 | *** join/#asterisk Posterdati (~Posterdat@95.238.224.91) |
| 15:09.20 | *** join/#asterisk Posterdati (~Posterdat@95.238.224.91) |
| 15:09.46 | Posterdati | [TK]D-Fender: so incoming calls now works |
| 15:30.49 | Posterdati | ok |
| 15:30.54 | Posterdati | thanks for help |
| 15:30.56 | WHiZZi | disable sip alg on the Cisco . |
| 15:31.10 | Posterdati | I did it too! |
| 15:31.53 | Posterdati | no ip nat service sip udp port 5060 |
| 15:32.09 | WHiZZi | ok. |
| 15:32.16 | Posterdati | no ip nat service sip tcp port 5060 |
| 15:32.29 | Posterdati | the latter is wrong on my router! |
| 15:32.34 | Posterdati | let me check |
| 15:35.08 | Posterdati | WHiZZi: I have only no ip nat service sip udp port 5060 |
| 15:36.08 | WHiZZi | that's fine, most SIP-traffic goes by UDP anyway |
| 15:36.30 | Posterdati | WHiZZi: ok |
| 15:36.44 | Posterdati | WHiZZi: shall I put back externaddr= ? |
| 15:36.59 | [TK]D-Fender | yes |
| 15:37.12 | [TK]D-Fender | It's not like we didn't tell you to do this a dozen times |
| 15:38.03 | Posterdati | the peer is back to be unreachable |
| 15:38.25 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:39.16 | Samot | JFC. |
| 15:39.29 | Samot | Stop putting the externaddr in there if you have a dynamic IP. |
| 15:39.32 | Samot | Just stop doing that. |
| 15:40.13 | Posterdati | ok |
| 15:40.32 | [TK]D-Fender | we can fix to externhost after |
| 15:40.43 | [TK]D-Fender | because that's what you should be using along with a DynDNS service |
| 15:41.42 | WHiZZi | My guess, without knowing the backstory here, try putting nat=force_rport in the sip.conf |
| 15:43.34 | Posterdati | WHiZZi: now I can receive calls, but cannot place a call from the sip phone, the question is this: shall I use the pbx as the provider for the sip phone? |
| 15:44.42 | [TK]D-Fender | ... |
| 15:44.59 | WHiZZi | yes |
| 15:45.43 | Posterdati | ok, what is the domain config for the phone? I mean in the phone panel! |
| 15:45.58 | WHiZZi | your pbx |
| 15:46.09 | [TK]D-Fender | Why are we not looking at the CALL? |
| 15:46.10 | Posterdati | nice |
| 15:46.59 | Posterdati | [TK]D-Fender: because the log remain empty as I place the call from the phone! |
| 15:47.15 | [TK]D-Fender | then you've screwed up something else |
| 15:47.22 | WHiZZi | because you registered your external sip-account on the phone? |
| 15:47.24 | Samot | [Mar 8 15:55:03] WARNING[-1][C-00000000]: chan_sip.c:24054 handle_response_invite: Received response: "Forbidden" from '"sip-phone" <sip:+390624405245@telecomitalia.it>;tag=as2cfca9c8' |
| 15:47.24 | Samot | <PROTECTED> |
| 15:47.30 | Samot | That was from a while ago. |
| 15:47.47 | Samot | So the phone has been a bit messy. |
| 15:47.58 | Posterdati | Samot: exactly |
| 15:48.18 | [TK]D-Fender | So how did you screw it up where before it was at least sending CALLS to your server? |
| 15:49.05 | Posterdati | well I tried to update the firmware to see if they fixed the 32 characters password problem... |
| 15:49.23 | Posterdati | then I clear the config and redo it |
| 15:49.49 | Posterdati | Samot: the phone is a bit messy alas it register to the pbx |
| 15:50.06 | WHiZZi | where are you registering your phone on? On your local pbx or on the sip-provider? |
| 15:50.19 | Samot | Those phones actually come pre-interop'd with like 6 Italian providers. |
| 15:50.20 | Posterdati | pbx |
| 15:50.28 | Samot | You have to pick the one they didn't interop with. |
| 15:50.34 | Posterdati | Samot: not with tim :( |
| 15:51.00 | Samot | WHiZZi: He ordered phone service, got a Gigaset phone, the phone doesn't support the password length the provider uses. |
| 15:51.03 | Posterdati | Samot: an operator from the provider, told me that gigaset AS540IP was ok! |
| 15:51.17 | Samot | WHiZZi: So this has been the last week. Getting Asterisk to act as the PBX... |
| 15:51.20 | Posterdati | Samot: it wasn't! |
| 15:51.42 | Samot | WHiZZi: For a single voip account. |
| 15:52.00 | Samot | That uses a single phone DECT device. |
| 15:52.17 | Posterdati | no 4 dects phones registerd on the base |
| 15:52.24 | Samot | OK.. |
| 15:52.26 | Samot | Whatever. |
| 15:52.26 | Posterdati | two AS540H |
| 15:52.35 | Posterdati | and two AS120 |
| 15:52.39 | Posterdati | :) |
| 15:52.43 | Samot | You basically signed up for simple resi/soho level voip service. |
| 15:52.58 | Samot | That should allow any dumb DECT/ATA device be used.. |
| 15:52.59 | Posterdati | a single phone at the time |
| 15:55.08 | Posterdati | Samot: the absurd is that I cannot use another provider, |
| 15:55.21 | Samot | Yes, it is. |
| 15:55.42 | Posterdati | Samot: because the line is owned by telecom and they didn't lend the line to other provider... |
| 15:55.53 | Samot | If this is a business and they decided paying you to be on IRC for a week to figure this out was cheaper then getting a better phone set or another provider..they are wrong. |
| 15:56.05 | Samot | And you can't port or move your number? |
| 15:56.11 | Samot | Italy doesn't let you change providers? |
| 15:56.24 | Posterdati | Samot: I can sign a contract with others but I have to always pay the line to telecomitalia |
| 15:56.42 | Samot | OK so you're stuck in a contract. |
| 15:56.46 | Posterdati | Samot: the do not mantain the number |
| 15:56.56 | Samot | But you won't buy a phone set that will work properly? |
| 15:57.02 | Samot | ..... |
| 15:57.03 | Posterdati | Samot: no, I'm stuck in this part of the city |
| 15:57.47 | Posterdati | Samot: I thought that sip phones would be standard and the provider too, plus the telecomitalia told me that it was ok |
| 15:57.59 | Samot | OK |
| 15:58.02 | Samot | They were wrong. |
| 15:58.09 | Posterdati | yes |
| 15:58.17 | WHiZZi | Posterdati: what's the goal in the end? What are you trying to accomplish here? |
| 15:58.17 | Samot | So now you're stuck with this provider. |
| 15:58.24 | Samot | Basic voip service. |
| 15:58.26 | Samot | Like Vonage. |
| 15:58.29 | Posterdati | who could know that in advance??? |
| 15:58.30 | Samot | That's what he got. |
| 15:58.50 | Posterdati | Samot: yes |
| 15:58.53 | Samot | Posterdati: What came firs the provider or the phone? |
| 15:59.04 | Posterdati | Samot: provider |
| 15:59.04 | Samot | Which one was purchased first? |
| 15:59.15 | Samot | OK so did you get all their details?\ |
| 15:59.22 | Samot | Did you know they have a 64 character password? |
| 15:59.25 | Posterdati | because I had an old adsl with them |
| 15:59.34 | Posterdati | with analog phones and filters |
| 15:59.39 | Samot | They provided you with your SIP account details. |
| 15:59.50 | Samot | You then went and got a phone that didn't support 64 character passwords. |
| 16:00.11 | Posterdati | Samot: NO, they do not know that so I complained with gigaset and them, telecom told me that I have to choose their damned modem |
| 16:00.18 | Samot | You bought a phone that doesn't have your provider listed in it's Italy's supported providers list. |
| 16:00.55 | Posterdati | Samot: I told you, the telecom customer care told me that it is |
| 16:01.01 | Samot | So? |
| 16:01.05 | Posterdati | it is not |
| 16:01.07 | Samot | A person who answers the phone... |
| 16:01.12 | Samot | At a low level... |
| 16:01.26 | Samot | Probably doesn't know the details of each model or maker out there. |
| 16:01.43 | Samot | That doesn't mean you shouldn't have done your own ressearch. |
| 16:01.45 | Posterdati | ok, but he could tell me where to look for a valid list of them |
| 16:01.57 | Posterdati | Samot: yes, I did an error |
| 16:01.58 | Samot | Dude, the answer is either learn networking and Asterisk.... |
| 16:02.05 | Samot | Or go buy a phone that works with them. |
| 16:02.33 | Samot | Like I said, if this is for a business you've already wasted a lot of money and time on this during this week. |
| 16:02.44 | Samot | And you're still not anywhere near being solved. |
| 16:02.50 | Posterdati | Samot: I think the phone could be used, the problem is the provider |
| 16:02.57 | Samot | OK |
| 16:03.03 | Samot | But you can't leave this problem. |
| 16:03.06 | Samot | So adjust. |
| 16:03.08 | Posterdati | Samot: no, it is not |
| 16:03.18 | Posterdati | Samot: no, it is not a business |
| 16:03.31 | Samot | OK..so you're just going to spend how much of your time on this? |
| 16:03.32 | WHiZZi | I think Posterdati needs to figure out what he wants, why is he dealing with Asterisk? |
| 16:03.39 | Samot | We know why |
| 16:03.43 | Posterdati | Samot: no, I won't put my hands in a job I do not know |
| 16:03.53 | Samot | His phone can't support 64 character SIP passwords. |
| 16:04.03 | Samot | His provider uses 64 character SIP passwords. |
| 16:04.12 | Samot | That's the entire issue, WHiZZi. |
| 16:04.21 | Posterdati | Samot: yes! |
| 16:04.21 | Samot | Asterisk is being used to get around that. |
| 16:04.27 | Samot | Which is completely dumb. |
| 16:04.44 | WHiZZi | ok, so his phones should register at asterisk |
| 16:04.48 | Posterdati | Samot: the problem is that I never used it before |
| 16:04.49 | WHiZZi | and asterisk to his provider |
| 16:04.59 | Posterdati | WHiZZi: more or less, yes! |
| 16:05.08 | Samot | WHiZZi: He has zero Asterisk skills. Along with zero networking knowledge. |
| 16:05.20 | Posterdati | Samot: not zero network |
| 16:05.28 | Samot | WHiZZi: If it hasn't been an Asterisk config issue, it's a router config issue. |
| 16:06.11 | Samot | WHiZZi: This is a 15 minute problem that he has turned into a week long problem. Mainly due to failure of following instructions. |
| 16:06.42 | WHiZZi | I figured |
| 16:07.24 | Posterdati | Samot: or your temper, which prevent you to be polite and effective in your explanations like externaddr one time I had to use it an another time not |
| 16:07.50 | Samot | You were told early not to use it |
| 16:07.54 | Samot | Then you kept putting it back |
| 16:07.58 | Samot | Why? I don't know. |
| 16:08.06 | Posterdati | BY WHO? |
| 16:08.12 | Samot | Me. |
| 16:08.13 | Posterdati | one told yes and another not |
| 16:08.39 | WHiZZi | Posterdati: Take one step at a time. 1) Register your phone(s) at the PBX and make sure they can call eachother. |
| 16:08.56 | WHiZZi | 2) Register your SIP-provider on the Asterisk |
| 16:09.06 | WHiZZi | 3) Make a dialplan |
| 16:09.19 | Samot | 9:23:54 AM <Posterdati> Samot: no, dynamic ip |
| 16:09.19 | Samot | 9:24:02 AM <Samot> Uhm.... |
| 16:09.19 | Samot | 9:24:10 AM <Samot> Then you can't set a static external IP |
| 16:09.24 | Samot | 9:24:17 AM <[TK]D-Fender> you can |
| 16:09.24 | Samot | 9:24:26 AM <[TK]D-Fender> and when it changes you'll be lying to the other side |
| 16:09.24 | Samot | 9:24:31 AM <[TK]D-Fender> which will be bad |
| 16:09.24 | Samot | 9:24:34 AM <[TK]D-Fender> but until then will work |
| 16:09.30 | Posterdati | WHiZZi: now... The system worked, but I had an issue: the incoming calls lasted 1 minutes... |
| 16:09.43 | WHiZZi | that's a NAT issue |
| 16:09.56 | WHiZZi | nat=force_rport,comedia |
| 16:09.56 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-128-59.satx.res.rr.com) |
| 16:10.01 | Posterdati | WHiZZi: which I solved |
| 16:10.22 | Samot | No, not if the calls are lasting 1 minute. |
| 16:10.26 | Samot | There's an issue. |
| 16:10.45 | Posterdati | WHiZZi: because now I have incoming calls to last until I hang up |
| 16:11.05 | WHiZZi | that's how it should work :P |
| 16:11.37 | Posterdati | WHiZZi: yes, but now I cannot place an outbound call :) and I'm trying to debug it |
| 16:12.05 | Posterdati | as Samot could see, the sip phone is messy now |
| 16:12.51 | Samot | Actually, I've been saying it from the beginning. |
| 16:12.51 | Posterdati | alas it registers there's something wrong |
| 16:13.02 | Samot | Because those Gigaset's are not great. |
| 16:13.03 | Posterdati | Samot: true |
| 16:13.08 | Samot | People always seem to have issues with them. |
| 16:13.13 | Posterdati | Samot: they are cheap |
| 16:13.20 | Samot | Right. |
| 16:13.26 | Samot | They aren't great. |
| 16:13.29 | WHiZZi | There's a reason for that |
| 16:13.34 | Samot | They are subpar. |
| 16:13.42 | Samot | You got a non-standard provider and a subpar device. |
| 16:13.49 | WHiZZi | especially the A-series |
| 16:14.04 | WHiZZi | S-series and N-series are pretty decent |
| 16:14.39 | WHiZZi | and business phones are reasonable too |
| 16:15.18 | WHiZZi | N300A or N510 as endpoint, S-series for DECT. Work fine with Asterisk |
| 16:15.31 | Posterdati | Samot: I would never use such things for business |
| 16:21.04 | Posterdati | Samot: consider that the provider wants 5 euros per month for the modem for all the contract time |
| 16:41.05 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 16:42.17 | *** join/#asterisk pchero_work (~pchero@87.213.240.121) |
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| 17:13.47 | drmessano | cccccckddrvffilkkhlijfckcifibnguecnrhdijhfgg |
| 17:13.55 | drmessano | This fucking keyboard' |
| 17:14.01 | file | hello Yubikey |
| 17:14.16 | danjenkins | ðð |
| 17:14.34 | file | your OTP is valid, congrats |
| 17:14.37 | drmessano | Damn laptop is too small |
| 17:15.52 | drmessano | I need a right-angle USB extender that puts this thing like 3 inches out to the left and further back lol |
| 17:24.26 | *** join/#asterisk bobjase (47ae5b95@gateway/web/freenode/ip.71.174.91.149) |
| 17:24.30 | bobjase | in my sip.conf, I have allowguest=no, but when I do "sip show channels" I see a bunch of random connections with a last message of "Rx: INVITE" |
| 17:24.47 | bobjase | Any recommendations? |
| 17:26.25 | drmessano | file: https://github.com/pallotron/yubiswitch/releases |
| 17:26.31 | drmessano | Not all heroes wear caps |
| 17:26.33 | drmessano | Not all heroes wear capes |
| 17:26.35 | drmessano | or caps |
| 17:28.13 | [TK]D-Fender | bobjase, ignore those. So they sent you an invite. Nowhere does that say you accepted it |
| 17:28.44 | [TK]D-Fender | if you don't see an actual active * channel "core show channels" then it's not actually in process. |
| 17:34.26 | bobjase | there are lots of them and it's annoying - there's no way to block them? |
| 17:35.05 | [TK]D-Fender | Go firewall everything except spcific IP's you care to hear anything from |
| 17:36.05 | [TK]D-Fender | either preemptively (default "disallow" firewall) or retroactively (fail2ban) |
| 17:41.52 | bobjase | we have fail2ban |
| 17:41.56 | bobjase | but it doesnt block these ppl |
| 17:45.07 | [TK]D-Fender | perhaps not configured properly |
| 17:45.21 | [TK]D-Fender | Or this is just the latest batch who will eventually get banned |
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| 20:20.10 | Ai9zO5AP | Hello |
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| 20:44.22 | Posterdati | Samot: hi! I was able to fix incoming and outbound calls |
| 20:46.55 | Samot | Yay. |
| 20:47.05 | Samot | And what was the fix? |
| 20:48.07 | Posterdati | directmedia=no in general, trunk and phone |
| 20:48.12 | Posterdati | plus |
| 20:48.29 | Posterdati | I used 5060 udp and 10000 20000 range for the sip phone |
| 20:49.04 | Posterdati | now I have to wait to see if asterisk is able to get an incoming call after some minutes it has received one |
| 20:50.12 | Posterdati | sometimes happen that options and invite are sent to the wrong ips |
| 20:51.06 | Posterdati | ah I put in general keepalive=yes |
| 20:53.48 | Posterdati | Samot: what is an Ericsson MTAS - CXP9020729/8 R14L01 ?? |
| 20:54.29 | Samot | That sounds like something you should ask Google. |
| 20:55.06 | Posterdati | Your search - Ericsson MTAS - CXP9020729/8 R14L01 - did not match any documents. |
| 20:55.45 | Posterdati | https://www.ericsson.com/en/portfolio/digital-services/cloud-communication/cloud-volte-and-evolved-communication/ims/multimedia-telephony-application-server |
| 20:56.07 | Posterdati | wasn't ericsson bankrupt? |
| 20:57.10 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
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| 23:58.54 | robink | Anyone know of any outstanding issues when linking pjproject (2.8) against OpenSSL 1.1.0? |
| 23:59.21 | robink | I seem to be having difficulty, and I can't find any current (i.e. open) issues on the Asterisk/PJSIP bugtracker. |
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