| 00:11.10 | *** join/#asterisk elyob (~elyob@92.91.9.51.dyn.plus.net) |
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| 01:03.11 | *** join/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 01:03.58 | TheHonorableKitt | can someone do me the huge favor of explaining why you need to 'register'? If a certain user is given a name and password, isn't that a registration? Maybe I'm missing a definition here? |
| 01:04.40 | Samot | Register provides location services. |
| 01:05.14 | Samot | The endpoint is telling the PBX where to send callsm |
| 01:05.27 | TheHonorableKitt | is there any way to set asterisk to force apps to re-register then? users keep un-registering so voip calls are not possible |
| 01:06.00 | Samot | Thats the device |
| 01:07.13 | TheHonorableKitt | hmm alright |
| 01:07.15 | Samot | The device decides if and when to register and how long that location should be used |
| 01:08.02 | Samot | Are you having issues? |
| 01:08.17 | TheHonorableKitt | no, just what you said is what's happening :) |
| 01:08.36 | TheHonorableKitt | on a different note, where might I find the /contrib/scripts folder? |
| 01:08.51 | Samot | What are these users using? |
| 01:08.54 | TheHonorableKitt | trying to set this up: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial but I'm not sure where the location is for the second command |
| 01:09.07 | Samot | Because the device should register every X seconds |
| 01:09.26 | TheHonorableKitt | the 'Calls' android app |
| 01:09.55 | Samot | It should have a re-register setting |
| 01:10.23 | Samot | When the Expire time is reached, the device should send a new register |
| 01:10.51 | Samot | I am going to guess Calls is not backgrounding right |
| 01:11.46 | Samot | Do they end up as UNREACHABLE in the PBX? |
| 01:13.24 | TheHonorableKitt | nope, you're right, that user didn't have 'background services' enabled |
| 01:13.28 | TheHonorableKitt | i fixed it :) |
| 01:14.22 | TheHonorableKitt | samot, any thoughts on my other question with the link above? |
| 01:14.31 | Samot | Not backgrounding properly stops it from doing things |
| 01:15.03 | Samot | Dont use self sign |
| 01:15.35 | TheHonorableKitt | ? |
| 01:19.24 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 01:19.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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 |
| 01:19.28 | TheHonorableKitt | I use a self-signed cert for my website though, it's trusted just fine |
| 01:19.35 | Samot | No it is not. |
| 01:19.42 | TheHonorableKitt | :thinking: |
| 01:19.46 | Samot | If I go to your website, I will be told it is invalid. |
| 01:19.55 | Samot | And I would have to "Trust" the CA. |
| 01:20.14 | TheHonorableKitt | https://diaspsocial.com that doesn't show as trusted to you? |
| 01:20.14 | Samot | Once I add the CA to my trusted lists and download it...then future visits will not have that issue. |
| 01:20.40 | Samot | You're using Let's Encrypt. |
| 01:20.45 | Samot | They are a Public CA. |
| 01:21.00 | TheHonorableKitt | ah, then I'll just use that then :) |
| 01:21.19 | Samot | Then that's what you have to use. |
| 01:21.26 | Samot | diaspsocial.com |
| 01:21.34 | Samot | Because that is what the cert is for. |
| 01:22.00 | Samot | So diaspsocial.com would need to point to the PBX |
| 01:22.11 | Samot | So you can use it in the device properly. |
| 01:22.23 | Samot | That's going to break your website. You need a cert for your sip stuff. |
| 01:22.39 | TheHonorableKitt | ohhhhhhh didn't think about that |
| 01:22.41 | TheHonorableKitt | that's right |
| 01:22.46 | Samot | Plus you are going to need to reissue that cert for that server. |
| 01:22.54 | Samot | Since the key and csr are generated from the server. |
| 01:23.02 | TheHonorableKitt | that's on a contab, no biggie :) |
| 01:24.06 | Samot | Now for the real question; Are you doing this just to see if you can do it? Or do you think you have a need for it? |
| 01:24.17 | TheHonorableKitt | right now just to see if I can do it |
| 01:24.35 | TheHonorableKitt | in the future I'm hoping to develop a viable chat service for diaspora* |
| 01:24.50 | TheHonorableKitt | utilizing xmpp and asterisk |
| 01:24.59 | TheHonorableKitt | if at all possible, i'd rather get rid of xmpp |
| 01:25.15 | TheHonorableKitt | but, that's down the road a ways, so not on my radar yet |
| 01:26.00 | Samot | Why would you use Asterisk as a chat service? |
| 01:26.46 | TheHonorableKitt | diaspora is a 'facebook replacement', so I want to offer a viable replacement for 'messenger', which has phone call functionality |
| 01:27.00 | Samot | OK but Asterisk is a horrible choice for that |
| 01:27.07 | TheHonorableKitt | why? |
| 01:27.10 | Samot | It's not a chat service. |
| 01:27.29 | TheHonorableKitt | can you send texts from one person to another? |
| 01:27.36 | Samot | That's not a chat service. |
| 01:27.40 | Samot | That is a short message system |
| 01:27.53 | TheHonorableKitt | ok. if it works it works |
| 01:27.57 | Samot | OK |
| 01:28.09 | Samot | It requires _each user_ to have an _endpoint_ |
| 01:28.21 | TheHonorableKitt | that can be scripted |
| 01:28.24 | Samot | It sends messages _to_ endpoints and receives them _from_ endpoints. |
| 01:28.27 | Samot | JFC. |
| 01:28.27 | Samot | OK |
| 01:28.33 | Samot | It is not the right solution. |
| 01:28.40 | Samot | It can be used for the voice part... |
| 01:28.51 | Samot | But it's not the right solution for a "messenger" replacement |
| 01:29.12 | Samot | You can't have a "Facebook" replacement and then use things that are not anywhere close to how Facebook does it |
| 01:29.23 | Samot | Or how any real platform like that would. |
| 01:29.31 | TheHonorableKitt | are you seriously getting this worked up over this? I don't *have* to use this as a 'messenger', I can keep it to the xmpp that's already setup and working |
| 01:29.35 | Samot | Asterisk is a B2BUA/Telephony kit. |
| 01:29.46 | TheHonorableKitt | and, I *did* say, this is a long way off, it's not even on my radar atm |
| 01:30.01 | Samot | You said you wanted to get rid of XMPP. |
| 01:30.20 | Samot | Then when I pointed how the limitations and it was a wrong choice, your reply was "If it works, it works" |
| 01:30.26 | TheHonorableKitt | immediately after that, I said, "but, that's down the road a ways, so not on my radar yet" |
| 01:30.42 | Samot | OK and when it does get on your radar..... |
| 01:31.04 | Samot | You can start with the fact that Asterisk is not a proper solution for a chat service and not waste time going down that avenue. |
| 01:31.12 | TheHonorableKitt | 1. why the hell do you care? 2. why is this a conversation anyway, it's not even what I was wanting to talk about? 3. you're way too overworked about something that doesn't even matter |
| 01:31.22 | Samot | OK. |
| 01:31.28 | *** join/#asterisk petris (quassel@2600:3c02::f03c:91ff:fe25:b576) |
| 01:33.01 | Samot | 1. I really don't care. You asked for help on things. Based on your level of question and knowledge in those questions. I was trying to help you further by explaining things. 2. You brought it up. 3. I'm not overworked. |
| 01:33.26 | TheHonorableKitt | can we move on? I don't want to upset you, you don't want to upset me, let's just drop it, please |
| 01:33.44 | Samot | Sure? |
| 01:33.49 | Samot | Again, I'm not upset. |
| 01:34.43 | TheHonorableKitt | in order to get a cert , do I need to have a domain setup with DNS, etc.? if that's the case, I might as wel just use let's encrypt, but if not, I would just grab one from whereever |
| 01:35.16 | Samot | OK, I'm going to leave this conversation in general. |
| 01:35.47 | TheHonorableKitt | I'm talking about just encrypting my phone calls...... |
| 01:36.00 | Samot | I know. |
| 01:37.59 | Samot | Seriously though....... |
| 01:38.11 | Samot | Do you think you need a domain with DNS setup for this? |
| 01:38.23 | Samot | To get a cert and have it on a _working_ domain? |
| 01:38.45 | TheHonorableKitt | right now I'm just testing the feature, so it's not attached to a domain |
| 01:38.51 | Samot | You can't. |
| 01:39.02 | TheHonorableKitt | so then that answers the question |
| 01:39.09 | Samot | That's not how TLS certificates work. |
| 01:39.17 | Samot | They need a "Common Name" |
| 01:39.28 | TheHonorableKitt | so I'll just use my domain then.... |
| 01:39.32 | Samot | Again.. |
| 01:39.50 | Samot | Wow. |
| 01:40.38 | Samot | I guess he'll have fun when either his PBX doesn't work with the TLS or his website stops working... |
| 01:42.08 | Samot | Well I guess he'll be "The next Facebook" in regards to data/security breaches. |
| 01:42.23 | *** join/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 02:23.05 | TheHonorableKitt | anyone know what this might be? |
| 02:23.07 | TheHonorableKitt | [Nov 11 02:22:36] NOTICE[30037][C-00000000]: chan_sip.c:26407 handle_request_invite: Call from '' (216.244.84.218:5078) to extension '9011972595155517' rejected because extension not found in context 'default'. |
| 02:23.07 | TheHonorableKitt | localhost*CLI> |
| 02:23.25 | [TK]D-Fender | exactly what it says |
| 02:23.40 | TheHonorableKitt | so, someone from that IP is trying to use my sip service? |
| 02:23.47 | [TK]D-Fender | You are accepting a call from them and trying to process what they dialed in the context you clearly see it targeting |
| 02:24.00 | [TK]D-Fender | You allowed them in auth-wise |
| 02:24.04 | [TK]D-Fender | and they are hitting your dialplan |
| 02:24.18 | [TK]D-Fender | stop allowing un-authed calls |
| 02:24.23 | TheHonorableKitt | ah ok |
| 02:24.35 | TheHonorableKitt | where do I put that setting at, if you don't mind me asking please :) |
| 02:24.45 | [TK]D-Fender | in your PJSIP config clearly |
| 02:24.49 | [TK]D-Fender | go read the guides on it |
| 02:24.54 | [TK]D-Fender | and the sample config |
| 02:24.56 | TheHonorableKitt | yep, thanks man |
| 02:25.18 | [TK]D-Fender | I don't know the parameter name offhand as I haven't really used it personally. |
| 02:25.42 | TheHonorableKitt | any way to just disable pjsip entirely? I'm using chan_sip |
| 02:25.57 | TheHonorableKitt | oh, looks like that's on chan_sip anyway based on the log |
| 02:28.17 | *** join/#asterisk K0HAX (~michael@gateway/tor-sasl/k0hax) |
| 02:28.33 | Samot | 9:24:45 PM <[TK]D-Fender> in your PJSIP config clearly <-- PJSIP does not automatically allow "anonymous", just FYI. There is no PJSIP variation of "allowguests". |
| 02:28.48 | Samot | [TK]D-Fender: ^^^ |
| 02:28.54 | TheHonorableKitt | i'm not using pjsip |
| 02:29.00 | Samot | I wasn't speaking to you. |
| 02:29.10 | TheHonorableKitt | i love you too :* <3 |
| 02:30.48 | Samot | In your case, the proper setting in the [general] context would be allowguests=no |
| 02:31.26 | TheHonorableKitt | thanks :) I just set it |
| 02:31.44 | TheHonorableKitt | figured I'd just block that IP while I'm at it as well, don't need spam like that :P |
| 02:31.58 | Samot | iptables is your friend. |
| 02:32.05 | TheHonorableKitt | +1 |
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| 17:29.37 | \dev\cache | Hello, can someone please recommend a good GSM gateway rackmount with 8+ channels and support for SMS via, webui or some other software? |
| 18:23.01 | igcewieling | I suspect such a thing doesn't exist, except for maybe on high end devices. Even Sangoma has discontinued their GSM telephony cards. |
| 18:30.16 | \dev\cache | igcewieling: google says otherwise :) |
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| 18:54.26 | igcewieling | \dev\cache: how much for 8 gsm ports? |
| 19:15.33 | *** join/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 19:15.42 | TheHonorableKitt | when trying to setup mailboxes...should I be hearing something on the other end? like, something asking me to put in the extention number, and teh pin, etc.? or do I need to set that up myself? |
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| 19:28.38 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 19:29.45 | Samot | You need to send the call to the voicemail app |
| 19:35.03 | TheHonorableKitt | right, the *CLI says it's playing a sound, but I don't hear any sounds when I call it at all |
| 19:35.34 | [TK]D-Fender | then you probably have a config/networking issue |
| 19:35.50 | TheHonorableKitt | hmmmm |
| 19:35.51 | [TK]D-Fender | And need to look at the device you're communicating with specifically as well as your server |
| 19:43.08 | TheHonorableKitt | I'm not seeing anything wrong with the network setup :-$ It's not natted, my phone isn't on wifi so it's on mobile network, voicemail.conf is setup correctly, sip.conf is as well as extensions.conf. am I missing a config file here somewhere? |
| 19:43.25 | Samot | Show a debug |
| 19:45.00 | TheHonorableKitt | one sec let me pastebin it |
| 19:45.19 | TheHonorableKitt | https://pastebin.com/QkaVfvef |
| 19:46.33 | TheHonorableKitt | I called from my cell phone the voicemail number (configured as extension 6500), as you can see in the log, it says it's playing these audio files, but I don't hear any on my phone. I see no errors, it's as if the packates just aren't making it, as Fender said above, it looks like it should be a conf issue, but I'm not seeing anything that's wrong |
| 19:53.48 | [TK]D-Fender | you need SIP debug to prove where things are negotiated |
| 19:53.56 | [TK]D-Fender | "sip set debug on" |
| 19:54.07 | [TK]D-Fender | for both sides |
| 19:54.28 | TheHonorableKitt | ah, gotcha, ok, let me try again then |
| 19:55.08 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 20:00.25 | TheHonorableKitt | sorry taking so long, It's taking a minute to remove my ip's from the log |
| 20:00.47 | [TK]D-Fender | Don't |
| 20:00.53 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 20:00.55 | [TK]D-Fender | that is literally screwing with the evidence |
| 20:01.24 | TheHonorableKitt | yeah I'm sorry I'm not about to post my public ip address to a logged irc chat with 177 strangers. |
| 20:01.30 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:02.34 | [TK]D-Fender | Then do it privately. And learn about how to secure your system |
| 20:02.47 | [TK]D-Fender | if you're exposed to the internet you're going to end up getting scanned at some point |
| 20:06.03 | TheHonorableKitt | how about this, I'll replace the ip's with an identifying word (device-ip and server-ip). Would that do? |
| 20:08.37 | TheHonorableKitt | https://pastebin.com/yCMVXsiJ |
| 20:08.42 | TheHonorableKitt | There it is |
| 20:09.13 | TheHonorableKitt | I do notice an unknown IP address getting a 403 forbidden error. tried to do a lookup on it, but no information and lookup failed. I'm ignoring it, but included it in the log since it was in the middle of it all |
| 20:12.20 | Samot | OK so are you calling in remotely? |
| 20:12.29 | Samot | Do you have a phone that is remote to the PBX? |
| 20:12.47 | TheHonorableKitt | I'm calling in remotely from my mobile cell phone, yes |
| 20:13.00 | TheHonorableKitt | my asterisk setup is on a linode ubuntu server |
| 20:13.38 | Samot | So what is 9001? |
| 20:14.14 | Samot | Are you using an app on your mobile phone or are you calling a real DID over the PSTN? |
| 20:14.23 | TheHonorableKitt | i have three accounts on it at the moment: 9001 = me, 9002 = my wife, 9003 = my pc |
| 20:14.47 | TheHonorableKitt | i'm calling with an app on my phone, same issue seems to happen when calling with an app from my computer |
| 20:14.59 | Samot | OK. |
| 20:15.09 | Samot | So then that makes it easier to find this |
| 20:15.20 | Samot | o=- 3750954875 3750954876 IN IP4 192.168.1.177 |
| 20:15.40 | Samot | You're telling your PBX to send RTP to the devices LOCAL IP, which it can't do over the Internet. |
| 20:15.58 | TheHonorableKitt | O.o oh, strange. wonder where that setting is |
| 20:16.09 | Samot | Are they all using the same app? |
| 20:16.40 | TheHonorableKitt | no, one is the 'Calls' app on android, and one is the microsoftSIP app for windows |
| 20:17.25 | Samot | What kind of router do you have at your house? |
| 20:18.07 | TheHonorableKitt | I don't think that's the issue. Because both calls were from my phone, one was from while I was on wifi, and the other was off wifi using 3G cell service |
| 20:18.26 | Samot | Did you show the one on the wifi or on 3g? |
| 20:18.33 | TheHonorableKitt | that was a wifi one |
| 20:18.40 | TheHonorableKitt | let me do another with all the logs for you from 3g |
| 20:18.45 | Samot | OK, so in the case of what you have presented the router could be the issue. |
| 20:19.37 | TheHonorableKitt | strange it should do that, should just be a straight passthrough, but I'm not about to edit that router (it's a pfsense firewall that's very finicy) so let me get the test for you from 3G and see what comes from that |
| 20:20.23 | Samot | Oh boy |
| 20:20.45 | Samot | pfsense is notorious for screwing up SIP. |
| 20:20.54 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 20:25.04 | TheHonorableKitt | ok, here's the one from 3G: https://pastebin.com/V3LR9W9W |
| 20:25.32 | TheHonorableKitt | 6500 is the voicemail number, 9001 is the extention for my account that I'm calling from and also the mailbox I'm trying to gain access to |
| 20:25.48 | TheHonorableKitt | calling from the 'Calls' android app as well |
| 20:30.26 | Samot | o=- 3750956531 3750956532 IN IP4 100.109.209.225 |
| 20:30.32 | Samot | That's the IP it would be sending media too |
| 20:30.45 | TheHonorableKitt | O.o no idea what ip that is |
| 20:30.50 | TheHonorableKitt | but it looks like a private ip |
| 20:31.06 | Samot | Uhm. No. |
| 20:31.19 | Samot | If that is over 3g, then it would be the mobile carriers IP. |
| 20:31.34 | TheHonorableKitt | is there a way to set asterisk to only send media to the originating caller? |
| 20:31.44 | Samot | It IS |
| 20:31.50 | Samot | You're on the 3G network. |
| 20:32.00 | Samot | Therefore you're connecting over that network. |
| 20:32.03 | TheHonorableKitt | so it's being blocked by verizon? |
| 20:32.08 | Samot | I don't know. |
| 20:32.15 | Samot | rtp set debug on |
| 20:32.17 | TheHonorableKitt | cuz that IP isn't my phone's ip |
| 20:32.19 | TheHonorableKitt | ok |
| 20:32.30 | Samot | Well I don't know, you blocked it |
| 20:32.40 | Samot | So I can't compare and say "hey this is not matching as it should" |
| 20:32.56 | Samot | Part of TK's statement of "you're messing with the evidence" |
| 20:33.13 | TheHonorableKitt | holy jesus |
| 20:33.31 | TheHonorableKitt | ok I'll just send the pastebin to you privately then, is that ok? |
| 20:33.59 | Samot | Sure. |
| 20:35.18 | Samot | But you understand that you are making paste to a public site and not just people in this room will be able to see it |
| 20:35.26 | Samot | Like via pastebin itself. |
| 20:40.12 | TheHonorableKitt | bah too many darn lines, can't copy them all |
| 20:41.03 | TheHonorableKitt | I'll just toss you a snipet, because it's pretty much the same all the way across |
| 20:44.03 | TheHonorableKitt | actually, I have to head out man, thanks for the help so far. It looks like it's just bouncing from that 100 ip address to my cell's ip address, but nothing is getting through to my phone itself |
| 20:44.14 | TheHonorableKitt | could be the app, dunno, I'll have to do some more testing |
| 20:46.02 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 20:48.30 | Samot | I'm going to say it's the app. |
| 20:48.34 | Samot | And pfsense |
| 21:28.17 | compi123 | Samot: Can you please elaborate why pfsense is "notorious for screwing up SIP"? |
| 21:29.00 | compi123 | Samot: Is it for not being able to be configured well? |
| 21:29.39 | compi123 | Samot: Or is it because users do not configure it well for SIP? |
| 21:30.26 | Samot | Im driving right now...just at a light |
| 21:30.38 | Samot | In 20 mins sure |
| 21:30.57 | compi123 | Samot: :) |
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| 21:57.47 | igcewieling | https://support.onsip.com/hc/en-us/articles/204029430-PFSense-Firewall-Settings-for-VoIP maybe this too https://www.3cx.com/docs/pfsense-firewall/ |
| 22:00.12 | jamesaxl | compi123: you should be familiar with pfsense |
| 22:17.45 | Samot | compi123: It's notorious because it does. The "SIP Helpers", Port Randomization, poor NAT handling. |
| 22:19.01 | compi123 | Samot: Port random is not good for SIP? |
| 22:19.54 | Samot | compi123: Not when you're devices are behind NAT, register with a PBX and say "Here's my IP _and_ Port to send requests to" and then pfsense "randomizes" it so it breaks your SIP location. |
| 22:22.39 | Samot | I've been doing this a very, very long time. There are firewall appliances that are a PITA when it comes to SIP. SonicWall, ortigate, WatchGuard, FireBox, etc. At low "SIP loads" they tend to be fine but the more traffic they have they more they require a little TLC and tweaking.... |
| 22:22.45 | compi123 | Samot: I tried Pfsense a while ago. Then I set up a gateway with PF on FreeBSD. |
| 22:23.15 | Samot | But for pfsense, the best solution I have found for it is to replace it. |
| 22:23.45 | compi123 | Samot: I just built Asterix16 from source ready for configuration on FreeBSD. Any hints? |
| 22:23.59 | Samot | Nope. I don't use FreeBSD. |
| 22:25.27 | compi123 | Any hint for starting to configure? What to do first etc.? |
| 22:27.54 | Samot | As far as? |
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