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| 02:44.37 | *** 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 |
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| 06:49.01 | fnj00 | I am working on setting up 3-Way calling using a custom dialplan in extensions_custom.conf. I am running asterisk 13.22.0 through FreePBX. The issue I am experiencing is I first transfer my existing call using ChannelRedirect which works perfectly fine and the callie is transferred into the context which creates the MeetMe conference. I then collect the next number to be dialed using Read() and then pass that variable over to a Dial() |
| 06:49.01 | fnj00 | command in the same context. The call works and rings the other party. However once the other party answers the call ends and a hangup is triggered on the caller end. The other party remains connected to the call and the context stops at that point. |
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| 06:49.41 | fnj00 | Is there something specific I need to do when initiating a Dial() to ensure once the call is answered it remains connected. Is it related to the earlier ChannelRedirect() |
| 06:50.16 | fnj00 | where my initial channel is still active and the server fails to join the caller to the new callie? |
| 06:58.58 | fnj00 | This is my extensions_custom.conf, there are a number of commented out lines in there from past tests using Originate() over Dial() which would also work until the calle answered with the same hangup result on the caller side. |
| 06:59.01 | fnj00 | https://pastebin.com/L8X8yHfa |
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| 08:25.54 | TheGallopingFox | WLAN > OPENVPN > ASTERISK registers sip phone OK |
| 08:26.08 | TheGallopingFox | WAN > OPENVPN > ASTERISK doesnt register sip phone |
| 08:26.12 | TheGallopingFox | strange |
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| 08:51.08 | [TK]D-Fender | Very little is "strange". Packets are packets. |
| 08:58.52 | TheGallopingFox | maybe pfsense does some kind of NAT on WAN |
| 08:59.00 | TheGallopingFox | which stops sip replys etc |
| 08:59.18 | [TK]D-Fender | pfsense FUCKS packets. |
| 08:59.22 | [TK]D-Fender | and is a known offender |
| 08:59.27 | TheGallopingFox | :-( |
| 08:59.27 | [TK]D-Fender | specifically port randomization |
| 08:59.33 | [TK]D-Fender | ^ |
| 08:59.35 | [TK]D-Fender | disable that |
| 08:59.39 | TheGallopingFox | oh ok |
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| 17:47.47 | adeel | i'm seeing a weird behavior when using a local channel and call files....essentially, it looks like the 1 call file is creating 2 call attempts |
| 17:48.17 | adeel | oh i think i know what the problem is |
| 17:52.27 | adeel | nope, it's still doing something weird....running * 11.7.0, and when using a local channel in a call file, it appears that the local channel is spawning 2 call attempts |
| 17:53.42 | [TK]D-Fender | if you say so.... |
| 17:57.05 | adeel | [TK]D-Fender: more like the asterisk CLI says so....https://pastebin.com/RMRWDuX0 |
| 17:58.16 | Samot | That's because Local channels are "two-leg" channels. |
| 17:58.34 | Samot | You create an outbound leg to make the call.... |
| 17:58.48 | Samot | Once it is answered, it "flips" the leg to inbound.... |
| 17:59.15 | adeel | Samot: yeah but i see 2 calls on the SBC when there should have only been one....and 2 sip channels in asterisk with 2 local channels |
| 18:00.27 | Samot | Well.... |
| 18:00.33 | file | the console output does not match your dialplan. |
| 18:00.46 | Samot | [Nov 10 12:51:07] -- Executing [1@incoming:1] Goto("Local/1@incoming-0000000b;2", "SSCoutbound,1,1") in new stack |
| 18:00.50 | Samot | Thats your first call |
| 18:01.11 | adeel | Samot: yup |
| 18:01.21 | Samot | And then when it is answered... |
| 18:01.27 | Samot | You've "looped" yourself. |
| 18:01.43 | adeel | it spawns another call after its been answered? |
| 18:01.51 | file | it does as it has been told to do |
| 18:01.52 | Samot | I would need to see your dialplan. |
| 18:02.02 | adeel | Samot: it's at the end of the pastebin |
| 18:02.41 | Samot | Show the call file |
| 18:03.32 | Samot | And what file said is very true. |
| 18:03.46 | Samot | Your dialplan show the priority 1 being commented out |
| 18:03.50 | Samot | And thus the dial() |
| 18:03.50 | file | the dialplan you posted is broken too, it lacks a priority 1 |
| 18:03.55 | Samot | But your log shows it being run |
| 18:04.00 | Samot | And there is that |
| 18:04.10 | Samot | By commenting it out, you've removed it. |
| 18:04.45 | Samot | So let's see the call file template. |
| 18:06.23 | adeel | here's the callfile template, https://pastebin.com/NcrwtaiJ |
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| 18:22.41 | [TK]D-Fender | you aren't defining a context in there |
| 18:22.49 | [TK]D-Fender | and you literally have the extension in there TWICE |
| 18:26.56 | fnj00__ | I have been attempting to setup n-way calling using MeetMe and contexts defined in /extensions_custom.conf. The first caller is redirected to the MeetMe conference no problem. Its when the second callee answers that the caller hangs up the call. My contexts are https://pastebin.com/L8X8yHfa. |
| 18:27.41 | fnj00__ | I wonder if its my use of Local/${EXTEN}@from-internal in that Dial() statement that may be incorrectly defined. |
| 18:28.09 | fnj00__ | The call rings through but as soon as the other end answers the caller terminates the call and the dailplan does not progress any further. |
| 18:28.16 | [TK]D-Fender | we don't see what your feature is doing |
| 18:28.23 | [TK]D-Fender | exten => s,n,Dial(Local/${DEST}@dynamic-nway-dest,,) |
| 18:28.40 | Samot | adeel: That call file is not correct. |
| 18:28.42 | [TK]D-Fender | and I'm thinking you are p[robably calling this from WITHIN a a currently active channel |
| 18:28.56 | [TK]D-Fender | doing another DIAL will cause its own bridge which is shooting yourself in the foot |
| 18:29.01 | fnj00__ | Yeah the channel is still active after the ChannelRedirect is made |
| 18:29.13 | [TK]D-Fender | If you want to spawn things you need to do that entirely external from the active channel |
| 18:29.40 | [TK]D-Fender | which means executing a backgrounded external script to launch that 2nd callusing a call-file or external script |
| 18:29.50 | [TK]D-Fender | I'm talking about the DIAL() you have in there |
| 18:30.05 | [TK]D-Fender | Show the feature and actual usage attempt |
| 18:31.36 | fnj00__ | features_applicationmap_custom.conf -> https://pastebin.com/dD0NWheR |
| 18:35.31 | [TK]D-Fender | looks like it might turf the transferee with the redirect and potentially continue on to prompt and dial the next |
| 18:35.38 | [TK]D-Fender | so go show what's actually happening |
| 18:36.12 | fnj00__ | yep grabbing the log now, im on a remote machine on windows so im a tad bit slower pulling that data out |
| 18:52.16 | fnj00__ | Pull of the logs during the attempt. Start the pull after the transfer of the first caller is made: https://pastebin.com/fYiBCHp3 |
| 18:55.56 | fnj00__ | Line 235 is where you see the answer and the bridge channel then the immediate exit of 2268 and proceeding hangup |
| 18:57.06 | fnj00__ | I also tried this with a originate and with that I was able to have the second caller transfer to the MeetMe but experienced the same issue with the hangup of the caller |
| 18:58.58 | fnj00__ | I got it to work with originate of the second caller and then tossing a MeetMe() at the end of the macro-nway-start context which you see commented out in there currently |
| 18:59.45 | fnj00__ | That brought all 3 callers into the MeetMe conference but is not ideal as it would also bring in a voicemail box if the 2nd caller transferred to voicemail instead of answering. |
| 19:00.53 | fnj00__ | My plan was to have the second call connect and then if the caller dials *42 then it would transfer the second caller into the conference and then complete the macro-nway-start context ending with the caller also transferring themselves over. |
| 19:02.08 | fnj00__ | Whereas if the caller dialed #42 it would end their call with the second callee and then transfer them into the MeetMe |
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| 19:14.04 | [TK]D-Fender | I'm thinking this is because the way you redirected the previous bridgepeer channel the origianl channel that continues on eventually wants to get "out" of that spawned sub-channel from the original feature trigger |
| 19:15.37 | fnj00__ | o[[[[[]'''''''' |
| 19:15.40 | [TK]D-Fender | I would rewrite this entire process. Have your feature call an external script that is backgrounded (essential). wait 1 sec so the channels re-bridge and the feature "resolves), then use an AMI Redirect (MUCH better) and redirect BOTH legs to separate dialplan so that they can continue without a counter hanging over their heads (feature timeout expecting a rebridge) |
| 19:16.11 | [TK]D-Fender | You could also be getting burned by local channel optimization since you aren't using /n on the end of them all and could be optimizing out of being able to continue each nest bit that may matter. |
| 19:16.30 | [TK]D-Fender | so I would totally avoid ALL of that and just make the trigger external so it's 100% clean |
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| 19:53.14 | *** join/#asterisk TheHonorableKitt (47ab630a@gateway/web/freenode/ip.71.171.99.10) |
| 19:53.23 | TheHonorableKitt | hey guys o/ |
| 19:54.37 | TheHonorableKitt | anyone on here able to help me out with an issue I'm having? First time setup for asterisk, following this tutorial: https://linuxhint.com/install_asterisk_voip_ubuntu/ When I try to connect via the CSIPSimple android app, I get 'Error while registering - Unauthorized' |
| 19:56.28 | fnj00 | How did you configure your auth settings? |
| 19:58.34 | TheHonorableKitt | exactly the same as the tutorial states, word for word |
| 19:59.41 | fnj00 | See anything in the full log? |
| 19:59.59 | TheHonorableKitt | I checked /var/log/asterisk and there's nothing in there |
| 20:00.07 | fnj00 | Did you reload asterisk after setting up the extensions? |
| 20:00.22 | TheHonorableKitt | i mean there's a few folders, one file, and the file is empty, as well as the folders |
| 20:00.31 | TheHonorableKitt | yes, I did restart asterisk after any edits I made |
| 20:02.19 | fnj00 | See anything through the asterisk cli? |
| 20:02.31 | fnj00 | Via asterisk -r |
| 20:02.42 | TheHonorableKitt | one sec |
| 20:02.52 | *** join/#asterisk lostinsip (~nick@82-71-3-134.dsl.in-addr.zen.co.uk) |
| 20:03.09 | TheHonorableKitt | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 20:03.22 | lostinsip | hello |
| 20:03.56 | fnj00 | Did you check if its running? |
| 20:04.35 | TheHonorableKitt | oof I thought I had to run that as a new session, so I stopped asterisk |
| 20:04.59 | lostinsip | I apologise in advance for any completely noob questions I may ask |
| 20:05.39 | TheHonorableKitt | ok I'm in the cli, but I don't see anything here?? |
| 20:05.54 | lostinsip | After several hours and various google searches later im running out of hair to pull so thought id try IRC |
| 20:06.10 | TheHonorableKitt | lostinsip please state your question :) |
| 20:06.25 | lostinsip | Thanks |
| 20:06.29 | fnj00 | core set verbose 17 |
| 20:06.38 | fnj00 | Then you should see verbose logs |
| 20:07.25 | TheHonorableKitt | ah hmmm "no matching endpoint found" and "failed to authenticate" |
| 20:08.10 | fnj00 | Check configs to ensure your endpoint is setup properly |
| 20:08.46 | fnj00 | core show endpoints |
| 20:09.16 | lostinsip | I have a small PBX running asterisk via FreePBX and so far its been fine, I have various softphones in use (2 instances of CsipSimple on android and 3 instances of Twinkle on linux) the issue is that the latest extension I have added (ext 105) despite all settings (as far as I can see) being identical on the client the extension will not register |
| 20:09.21 | fnj00 | core show extensions |
| 20:09.24 | lostinsip | I get a 401 error |
| 20:10.03 | lostinsip | I have checked the secret is correct, restarted the server, restarted the client |
| 20:10.32 | lostinsip | checked the details against the other extensions in the client and the server gui |
| 20:10.52 | fnj00 | Which gui lostinsip |
| 20:11.07 | lostinsip | FreePBX |
| 20:11.29 | fnj00 | Version? |
| 20:11.43 | TheHonorableKitt | I think I might know what's going on. It seems it's trying to use port 29312, but I only unblocked 10000-20000 |
| 20:11.55 | lostinsip | 14.0.4.5 |
| 20:12.03 | TheHonorableKitt | not sure why it's giving out this port hmmm, let me check my configs, thanks fnj00 |
| 20:13.13 | fnj00 | I think thats defaul for asterisk. Your client is trying the higher ports. Are you behind a nat or different vlan/ip range |
| 20:13.20 | lostinsip | I am happy to put any conf files or other data in a pastebin if it would help but im not sure which files would be most relevant |
| 20:13.52 | fnj00 | lostinsip is it configured for sip or pjsip |
| 20:14.19 | TheHonorableKitt | negative, I'm using a linode server with a public facing IP for my SIP server |
| 20:14.37 | lostinsip | configured for pjsip as I believe it is the newer version and to be preferred? |
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| 20:19.59 | TheHonorableKitt | well that's quite strange. rtp.conf is set to only use between 10000 and 20000, but it's giving ports outside of that range |
| 20:20.24 | fnj00_ | I don't think it's giving out the ports I think your client is trying to use those ports |
| 20:21.56 | fnj00_ | @lostinsip did you restart free PBX or reload after adding the latest extension |
| 20:23.39 | fnj00_ | Also verify you're using the secret for the extension as well as the extension to login using your client |
| 20:25.03 | lostinsip | I used the Apply Config button after adding the extension. refreshed the gui and then When it didn't work I tried restarting the machine the sever is running on, this was a software initiated restart from the terminal "reboot now" |
| 20:25.36 | lostinsip | I have checked the secret is correct (several times) |
| 20:26.17 | TheHonorableKitt | yep, you're right fnj00_, my wifi was choosing the wrong ports, strangely enough. Off wifi my phone connects properly, but I still see the same errors now, even with the correct port utilized |
| 20:30.26 | fnj00_ | Did core show endpoints show your endpoint configured? |
| 20:31.29 | TheHonorableKitt | localhost*CLI> core show endpoints No such command 'core show endpoints' (type 'core show help core show endpoints' for other possible commands) |
| 20:33.04 | TheHonorableKitt | should it be 'pjsip show endpoints' instead? |
| 20:36.38 | fnj00_ | Yes sorry mobile |
| 20:40.35 | TheHonorableKitt | hmmm, alright, i think I get endpoints now, I changed my configs to use 1101 and 1102 instead of 9001 and 9002 (as the tutorial I provided showed). So now the endpoint error isn't there, but now I just see: Failed to authenticate |
| 20:41.11 | TheHonorableKitt | [Nov 10 20:39:30] NOTICE[25050]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'REGISTER' from '<sip:1102@redacted>' failed for 'redacted:12035' (callid: HOz0jHVMtwl2GP6.BM72zFcBdbvh2WPP) - Failed to authenticate |
| 20:44.21 | fnj00_ | You verified your using the secret entered in your configuration for your password? |
| 20:44.26 | lostinsip | As an experiment I tried to register from another client as ext 105 as I wondered if it was an error on the client side, I set up a different user account in CsipSimple and that is failing to authorise as well, the original user account/ext on that device (101) registers fine |
| 20:45.22 | lostinsip | Yeah as this is currently a test PBX and not connected to the outside world I have used the same simple secret for all the devices |
| 20:45.57 | lostinsip | same setup procedure for all the extensions |
| 20:46.03 | fnj00_ | lostinsip does that endpoint show via asterisk cli > pjsip show endpoints |
| 20:47.20 | TheHonorableKitt | welp, something isn't right, I don't know what it is. I don't even know what the pjsip.conf is or what these endpoints are, they have their own auths and passwords and usernames, but none of this is making any sense to me |
| 20:47.29 | TheHonorableKitt | I'm royally and completely lost now smh :( |
| 20:48.30 | fnj00_ | Where are your extensions defined, which config file? |
| 20:48.39 | TheHonorableKitt | extensions.conf |
| 20:49.03 | fnj00_ | pastebin a example with secret redacted |
| 20:49.14 | TheHonorableKitt | one sec |
| 20:49.20 | lostinsip | this is where i will need some guidance, I have access to the sever as root in a linux terminal what command string do I need to enter to get the output i need, I have been using the FreePBX gui to configure up to now |
| 20:49.48 | lostinsip | I am familiar with the linux command line but not the asterisk one |
| 20:50.03 | fnj00_ | On free PBX that should actually show under asterisk info |
| 20:50.33 | fnj00_ | Should be a tab for PJSip endpoints to list them all |
| 20:51.50 | lostinsip | ok thanks i will look |
| 20:53.23 | lostinsip | The endpoint shows up |
| 20:53.32 | lostinsip | I can post a screenshot |
| 20:53.34 | TheHonorableKitt | Here's extensions.conf: https://pastebin.com/TaWJRFee |
| 20:53.43 | TheHonorableKitt | and here's sip.conf: https://pastebin.com/cHsxeJpw |
| 20:54.44 | TheHonorableKitt | and here's the output of pjsip show endpoints https://pastebin.com/RXfAv897 |
| 20:54.48 | fnj00_ | In FreePBX each extension has a section with the extension and secret then further down there is a user section with separate password. |
| 20:55.12 | TheHonorableKitt | i know, I changed that password to one of my own, and it still didn't authenticate |
| 20:55.27 | fnj00_ | Make sure you are using the secret just under the extension not the password defined under the user |
| 20:56.12 | fnj00_ | @thehonorablekitt those are sip endpoints |
| 20:56.27 | TheHonorableKitt | oh hmmm ok |
| 20:56.28 | fnj00_ | Is sip configured to listen on 5060 or 5160 |
| 20:56.33 | TheHonorableKitt | 5060 |
| 20:57.44 | TheHonorableKitt | where in my extensions.conf should I put the secret passwords? |
| 20:57.54 | TheHonorableKitt | should they be under their own brackets? |
| 20:58.33 | fnj00_ | Nope those go in sip.conf, extensions defines the ext numbers |
| 20:59.13 | TheHonorableKitt | so then what's wrong with my config? I'm confuse |
| 20:59.16 | TheHonorableKitt | confused* |
| 20:59.25 | fnj00_ | Usually bindport is defined in sip.conf, u sure its set to listen on 5060? |
| 20:59.55 | lostinsip | https://imgur.com/a/wiB86aU |
| 21:00.47 | lostinsip | I have only been using the password for the extension, its worked on the other setups |
| 21:01.09 | TheHonorableKitt | hmmm, alright, so the setting is 'bindport=5060'? |
| 21:01.16 | TheHonorableKitt | in sip.conf under [general]? |
| 21:01.35 | [TK]D-Fender | Show us the actual error |
| 21:01.55 | fnj00_ | Yep |
| 21:02.32 | [TK]D-Fender | You're pointing to configs and asking why it's failing but not looking at what is actually happening. |
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| 21:03.02 | [TK]D-Fender | You can point to brochure of your car, just don't go asking why it failed. We can't tell you put the car in reverse and crashed it into a wall. |
| 21:03.24 | fnj00_ | I think its pjsip listening on 5060 and your extensions are sip not pjsip |
| 21:03.58 | TheHonorableKitt | Fender, is that to me? |
| 21:04.50 | TheHonorableKitt | if it's to me, I already showed the actual error. #scrollbutton |
| 21:05.08 | lostinsip | Im going to eat some food back shortly |
| 21:05.51 | TheHonorableKitt | [Nov 10 21:03:19] NOTICE[17548]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'REGISTER' from '<sip:1107@redactedip>' failed for 'redactedip:12057' (callid: ZJj1uxc8ULmNcVg2u9CIBARD.CJrXhy5) - Failed to authenticate |
| 21:06.44 | TheHonorableKitt | that's after I set bindport=5060 on sip.conf |
| 21:07.30 | [TK]D-Fender | Guess what, they are FIGHTING over that port |
| 21:07.36 | [TK]D-Fender | and chan_sip lost |
| 21:08.09 | fnj00_ | Yeah that message is pjsip right at the beginning |
| 21:08.15 | TheHonorableKitt | Ok, what do I have to do to fix this then? |
| 21:08.24 | [TK]D-Fender | TheHonorableKitt> and here's the output of pjsip show endpoints https://pastebin.com/RXfAv897 <- see you're already starting by skipping looking at what t3h SERVER listens for vs what those ENDPOINTS want |
| 21:08.41 | [TK]D-Fender | go change the ports you bind in pjsip |
| 21:08.45 | fnj00_ | You need to go to the PJsip.conf and change the bind Port there |
| 21:09.58 | *** part/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:10.10 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:10.12 | [TK]D-Fender | oops |
| 21:10.37 | *** join/#asterisk fnj00_ (~androirc@2607:fb90:d8:e4bb:9371:75ac:ad64:472c) |
| 21:10.51 | fnj00_ | (fnj00_) You need to go to the PJsip.conf and change the bind Port there |
| 21:10.58 | TheHonorableKitt | so um, does someone have the default username and password for the 1107 endpoint in pjsip.conf they can toss me? |
| 21:11.05 | TheHonorableKitt | I changed it, think it was set to something specific for a reason |
| 21:12.42 | [TK]D-Fender | there is no such thing as "default" |
| 21:12.52 | [TK]D-Fender | your config is your own. |
| 21:12.57 | [TK]D-Fender | I have no such 1107 |
| 21:13.02 | TheHonorableKitt | ok |
| 21:13.17 | [TK]D-Fender | if you want to see what parameter syntax is you can look at the sample configs provided with * |
| 21:13.19 | TheHonorableKitt | now my phone just sits at 'registering' |
| 21:13.34 | [TK]D-Fender | And you are talking about what the PHONE see... and not looking at ASTERISK |
| 21:13.35 | TheHonorableKitt | doesn't actually fail or anything |
| 21:13.42 | [TK]D-Fender | Do you see the problem here? |
| 21:13.56 | [TK]D-Fender | Prove that it is coming IN. |
| 21:14.01 | [TK]D-Fender | And what it lands on |
| 21:14.52 | TheHonorableKitt | <---- = noob, i.e. I don't know how to do that, if you can, pretty please, explain how to do the things you ask so that I can get you the information you need to help me :) <3 |
| 21:15.02 | [TK]D-Fender | how to do that/D. |
| 21:15.06 | [TK]D-Fender | you we jsut looking in * CLI |
| 21:15.10 | [TK]D-Fender | you were THERE. |
| 21:15.14 | [TK]D-Fender | what is coming in NOW? |
| 21:15.57 | igcewieling | heh, in 2 days 93.115.28.168 and 77.234.46.144 where banned by fail2ban at 7 different FreePBX sites. I assume the sites are on similar IP, but I've not checked. |
| 21:16.00 | TheHonorableKitt | nothing is coming in |
| 21:16.28 | [TK]D-Fender | "sip set debug on" |
| 21:16.46 | TheHonorableKitt | do I put that in the CLI? |
| 21:16.49 | [TK]D-Fender | yes |
| 21:16.57 | TheHonorableKitt | localhost*CLI> sip set debug on No such command 'sip set debug on' (type 'core show help sip set' for other possible commands) |
| 21:16.59 | [TK]D-Fender | that will tell chan_sip to show what's coming in in full |
| 21:17.09 | [TK]D-Fender | then chan_sip isn't loading period |
| 21:17.18 | [TK]D-Fender | is that good? Are you now expecting pjsip ONLY? |
| 21:17.31 | TheHonorableKitt | I think I should start at the beginning. |
| 21:17.40 | [TK]D-Fender | I think you should tell us what you're expecting |
| 21:17.49 | [TK]D-Fender | if PJsip all we're supposed to care about now? |
| 21:18.01 | [TK]D-Fender | if so we can continue and move on. |
| 21:18.21 | [TK]D-Fender | "pjsip set logger on" |
| 21:19.04 | [TK]D-Fender | and go prove what your transport(s) are bound to |
| 21:19.48 | TheHonorableKitt | hang on, I'm trying to find the tutorial I used to install this, so you can see what I'm trying to do |
| 21:19.58 | TheHonorableKitt | I get that asterisk is a broad program, so I understand where your confusion might be |
| 21:20.18 | TheHonorableKitt | https://linuxhint.com/install_asterisk_voip_ubuntu/ |
| 21:20.34 | TheHonorableKitt | I set everything up, exactly, word for word, how this tutorial said. |
| 21:20.37 | [TK]D-Fender | I don't need a ton of histopry |
| 21:20.50 | TheHonorableKitt | right, but you asked what I'm trying to do, I'm trying to do that ^^ |
| 21:20.57 | [TK]D-Fender | are you trying to configure PJSIP for your device to connect to, or are you trying to configure chan_sip? |
| 21:21.05 | [TK]D-Fender | this is a simple question? |
| 21:21.25 | TheHonorableKitt | dude I don't know, I don't even know what pjsip is, or sip or extension, or whatever. I just followed the tutorial I literally have linked twice and i'm getting errors |
| 21:22.14 | [TK]D-Fender | * has to different channel drivers to speak the same language (SIP) |
| 21:22.28 | [TK]D-Fender | that guide explain only chan_sip (sip.conf) |
| 21:22.50 | [TK]D-Fender | yourprevisou CLI message with that error proved that chan_pjsip was actually listening on that port and chan_sip wasn't getting the request |
| 21:22.51 | TheHonorableKitt | ok, so how do I get sip.conf to work, and not pjsip.conf? |
| 21:22.58 | [TK]D-Fender | Now chan_sip isn't even loaded. |
| 21:23.07 | [TK]D-Fender | go fix the ports between the two of them |
| 21:23.13 | TheHonorableKitt | I already did. |
| 21:23.39 | TheHonorableKitt | neither initially had ports even bound or set, so I set them myself. |
| 21:24.11 | [TK]D-Fender | something was done wrong then |
| 21:24.15 | [TK]D-Fender | because chan_sip failed to load |
| 21:24.22 | [TK]D-Fender | "module load chan_sip.so" |
| 21:24.24 | [TK]D-Fender | see what happens |
| 21:25.25 | *** join/#asterisk mftech (~mftech@2a05:f6c3:3680:0:a476:469e:aa51:604b) |
| 21:26.36 | TheHonorableKitt | https://pastebin.com/umK7fusu |
| 21:26.44 | TheHonorableKitt | it's listening on 5060, as stated |
| 21:28.09 | TheHonorableKitt | -_- ooooook. well now it's working, but only after I ran 'sip set debug on' |
| 21:29.43 | [TK]D-Fender | You may not have been properly reloading |
| 21:29.54 | [TK]D-Fender | which is why starting the entire module worked now |
| 21:30.04 | [TK]D-Fender | it was never there for a basic "reload" to have any effect |
| 21:30.18 | TheHonorableKitt | yep, looks like it. guess I'll just have to run these commands every time my server reboots then |
| 21:33.19 | [TK]D-Fender | or maybe it failed to load because of a conflict that you hav now fixed |
| 21:33.23 | [TK]D-Fender | and a new start will happen properly |
| 21:33.32 | [TK]D-Fender | we don't actually know which will be the case yet |
| 21:35.05 | *** join/#asterisk mftech (~mftech@2a05:f6c3:3680:0:a476:469e:aa51:604b) |
| 21:35.57 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
| 21:38.02 | lostinsip | Back, sorry I had to eat my meal |
| 21:38.51 | lostinsip | I dont know if there is any insight into the issue I raised? I can always provide more information if needed |
| 21:40.13 | [TK]D-Fender | lostinsip, You need to be looking at PJSIP debug to prove what's coming in |
| 21:40.18 | [TK]D-Fender | "pjsip set logger on" |
| 21:40.42 | lostinsip | thanks, type that into the command line on the box? |
| 21:40.56 | [TK]D-Fender | * CLI |
| 21:41.21 | TheHonorableKitt | lostinsip, in another terminal with asterisk running, type 'asterisk -r' |
| 21:41.53 | lostinsip | ok i will SSH in |
| 21:42.02 | TheHonorableKitt | ok |
| 21:43.27 | lostinsip | so I have pjsip logging on |
| 21:45.05 | lostinsip | i tried to connect and got an error up I will post that |
| 21:45.07 | *** join/#asterisk mftech (~mftech@2a05:f6c3:3680:0:a476:469e:aa51:604b) |
| 21:46.16 | lostinsip | https://pastebin.com/57SLPL9b |
| 21:46.38 | lostinsip | this is output from the pjsip log when i hit register on the softphone linked to 105 |
| 21:46.43 | TheHonorableKitt | you might have the same issue I had |
| 21:46.50 | TheHonorableKitt | are you using pjsip.conf or sip.conf? |
| 21:47.34 | lostinsip | I am setting up the extensions through the GUI of FreePBX so i am not writing directly to the conf files |
| 21:47.44 | lostinsip | the setup is selected for pjsip |
| 21:47.54 | TheHonorableKitt | ok |
| 21:48.28 | lostinsip | I realise the GUI is writing the lines to a conf file but up till now the process has been pretty smooth |
| 21:48.55 | lostinsip | adding extensions, testing them setting up voicemail etc. all fine till 105 wont register |
| 21:48.59 | [TK]D-Fender | Go show us the client setup and the config setup for 105 |
| 21:49.11 | lostinsip | ok |
| 21:50.24 | lostinsip | For the config setup do you want a screenshot from the GUI or a copy of the data from the conf file? |
| 21:51.12 | [TK]D-Fender | anything at this point |
| 21:53.55 | lostinsip | ok i will put up the client screenshots first, |
| 21:55.14 | *** join/#asterisk mftech (~mftech@2a05:f6c3:3680:0:a476:469e:aa51:604b) |
| 21:58.10 | lostinsip | https://imgur.com/a/svNG3YG |
| 21:58.57 | lostinsip | setup pages from the twinkle softphone, set up the same as for 104 and 102 both of which work fine |
| 21:59.10 | lostinsip | I will do the conf file next |
| 22:02.50 | TheHonorableKitt | because I'm a total noob and don't fully understand how asterisk works, i have a question |
| 22:03.25 | TheHonorableKitt | is it possible to get the username and password from a postresql database for sip.conf? |
| 22:03.35 | TheHonorableKitt | and, protect said password so that it's not clear text? |
| 22:03.43 | [TK]D-Fender | "total newb" and "fully understand" are pretty much polar opposites.... |
| 22:03.52 | [TK]D-Fender | since we've established that... go for it :) |
| 22:05.00 | [TK]D-Fender | you can specify your configs to be pulled from SQL using "realtime". Look it up in the book |
| 22:05.04 | [TK]D-Fender | ~book |
| 22:05.04 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:05.12 | [TK]D-Fender | or the official wiki |
| 22:05.13 | [TK]D-Fender | ~asteriskwiki |
| 22:05.13 | infobot | somebody said asteriskwiki was http://wiki.asterisk.org |
| 22:05.40 | [TK]D-Fender | As for that setting being "not cleartext", yes, there is a parameter for an MD5hased secret |
| 22:06.17 | TheHonorableKitt | nice, ok, so long as I know it's possible, I'll figure it out eventually |
| 22:06.24 | lostinsip | Its probably easier for me to paste the conf file directly rather than try and screenshot the GUI which conf file is relevant? pjsip.endpoint.conf seems the most relevant to my untrained ey |
| 22:06.30 | lostinsip | *eye |
| 22:06.50 | TheHonorableKitt | I run a diaspora* server with about 500+ users, looking to potentially setup a voice chat service with asterisk to tie into it |
| 22:07.30 | [TK]D-Fender | Go read up on realtime and you'll have to make sure there is a completely properly formatted table for * to read to get its settings |
| 22:08.00 | lostinsip | sounds a lot more ambitious than my plan for a small business phone system TheHonorableKitt! |
| 22:08.01 | TheHonorableKitt | sounds fun :D |
| 22:08.23 | TheHonorableKitt | nah, shouldn't be that hard to setup |
| 22:09.11 | TheHonorableKitt | as long as I can echo the usernames into sip.conf and extensions.conf and md5 hash the passwords, all that's needed is a good web-browser sip client that I can tie into the chat service, and I already found that: http://jssip.net/ |
| 22:09.41 | lostinsip | chat/messaging isnt something i have tried yet |
| 22:10.08 | TheHonorableKitt | diaspora's chat service uses an XMPP service, so I do have that setup, not many folks use it |
| 22:10.34 | lostinsip | all the softphones have a messaging tab but the messaging doesnt work but I want to get the calling working first, this glitch with 105 is such a pain |
| 22:11.47 | TheHonorableKitt | I can confidently say that CSIPSimple android app is crap, doesn't even make the calls. I used an android app called "Calls", that one works great |
| 22:11.47 | lostinsip | https://pastebin.com/PSAyJf0A |
| 22:12.18 | lostinsip | Oh that sounds interesting, however i have had no issues with CsipSimple yet |
| 22:12.46 | TheHonorableKitt | your issues are definitely different than mine were, so I'm sorry, I really can't help. :( |
| 22:13.00 | lostinsip | that paste is the pjsip.endpoint.conf |
| 22:13.35 | [TK]D-Fender | And we're not seeing the auth |
| 22:13.40 | lostinsip | I can paste any others if needed maybe [TK]D-Fender may have more insight |
| 22:13.59 | [TK]D-Fender | clearly there is another important file "include"d |
| 22:16.01 | *** join/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 22:16.33 | *** part/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 22:17.54 | lostinsip | I have a large array of conf files, for pjsip some are empty pjsip.endpoint_custom.conf is empty for example |
| 22:18.11 | [TK]D-Fender | Of course "custom" is empty |
| 22:18.17 | [TK]D-Fender | that's only for you to fill in manually |
| 22:18.27 | Samot | Where's a real debug? |
| 22:18.33 | *** join/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 22:18.39 | Samot | Something that isn't just one SIP packet of a transaction? |
| 22:18.42 | lostinsip | Thats what i figured, its not written to by the GUI |
| 22:19.36 | lostinsip | I have compared the entries for the working extensions and 105 and so far cant see any difference |
| 22:19.59 | Samot | 5:18:26 PM <Samot> Where's a real debug? |
| 22:20.05 | Samot | 5:18:39 PM <Samot> Something that isn't just one SIP packet of a transaction? |
| 22:20.35 | lostinsip | but im too new to Asterisk to know where to look for the reason |
| 22:20.47 | Samot | You just posted it. |
| 22:21.05 | [TK]D-Fender | Do it again and get MORE. |
| 22:21.11 | Samot | 4:46:15 PM <lostinsip> https://pastebin.com/57SLPL9b |
| 22:21.17 | Samot | ^^That is incomplete. |
| 22:21.18 | [TK]D-Fender | you cut off the start of the register attempt |
| 22:21.27 | *** part/#asterisk TheHonorableKitt (~galat@pool-71-171-99-10.clppva.fios.verizon.net) |
| 22:21.44 | lostinsip | ok i will re-run |
| 22:26.38 | lostinsip | https://pastebin.com/GaCi4Hdc |
| 22:27.23 | lostinsip | thats all the log that came up from starting the logging fresh and from hitting register on twinkle |
| 22:28.29 | [TK]D-Fender | we haven't seen eith password in clearl |
| 22:28.38 | [TK]D-Fender | no auth from the config. Masked in the client |
| 22:30.33 | lostinsip | the client does not have a see password option |
| 22:31.13 | lostinsip | so i have to rely on correct typing |
| 22:31.35 | lostinsip | the password is quite simple tho and i have tried it several times |
| 22:32.29 | [TK]D-Fender | and we're not seeing the * side |
| 22:32.57 | lostinsip | sorry the * side? |
| 22:33.39 | [TK]D-Fender | Asterisk |
| 22:33.51 | lostinsip | oh of course |
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| 22:42.04 | lostinsip | I am sitting watching the log go past on the terminal, I can see my mobile register if i turn off the app and on again but I admit i have little more insight than that |
| 22:48.30 | lostinsip | I'm sorry I am going to have to give up for tonight its getting late. I will read more about Asterisk as I can and give another go at solving this tomorrow |
| 22:48.51 | lostinsip | Thanks for your help so far [TK]D-Fender and Samot |
| 22:50.11 | lostinsip | goodnight |
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