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00:39.00 | void09 | anyone around to help me set up my freepbx ? simple setup, dial in a SIP voip number (works) and then dial out through a SIM (set up as a SIP gateway, from a goip box) |
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01:40.00 | n0tiz | I'd like to wish everyone a happy new years! |
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12:24.57 | dongs | hmm, what can i do to debug peer to peer IAX2 connection? i've stopped asterisk and did nc -u -p 4569 and i can see the client trying to connect, i've done iax2 set debug on and i can see REGREQ and CTOKEN or wahtever, but client doenst seem to be registering. |
12:25.14 | dongs | i'm connecting android phone (iax2 softphone) -> asterisk by iax2. |
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12:34.37 | dongs | i actually don't understand what is the problem with IAX registration. i can see the incoming registration in iax2 debug but it doesn't generate a warning or notice or anything. |
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13:33.04 | void09 | Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
13:33.09 | void09 | goip32/goip 192.168.100.198 Yes Yes 5060 OK (2 ms) |
13:33.26 | void09 | this is what i get when i do sip show peers in asterisk console |
13:33.36 | void09 | 2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline] |
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13:43.18 | Samot | What is the actual problem |
13:45.25 | void09 | well, I have this in FreePBX |
13:45.44 | Samot | And? |
13:46.14 | void09 | https://pastebin.com/raw/1GqcmcTy |
13:46.30 | void09 | sip trunk |
13:46.45 | void09 | no dialed number manipulation |
13:46.54 | Samot | What? |
13:47.08 | Samot | You need to do that in dialplan |
13:47.13 | void09 | in sip trunk settings. there are no dialed number manipulation settigns |
13:47.22 | void09 | where is dialplan ? |
13:47.37 | Samot | That sip config is junk |
13:47.58 | Samot | Well FreePBX manages that |
13:48.27 | void09 | https://imgur.com/a/psm0bQC |
13:48.33 | void09 | this is my freePBX config |
13:49.17 | Samot | That is a screenshot of a trunk |
13:49.18 | void09 | i have another pjsip trunk that works, I can dial it. Inbound route is set to IVR |
13:49.20 | void09 | yes |
13:49.25 | void09 | I am show you my config |
13:49.26 | Samot | Just one tab |
13:49.40 | void09 | all 3 tabs |
13:50.16 | Samot | I see the tabs |
13:50.28 | Samot | Where are the screenshots |
13:50.39 | void09 | they're all 3 in the same link |
13:51.03 | void09 | in IVR I have digit 3 bound to Destination "DISA" |
13:51.13 | void09 | I enter my password, it's good |
13:51.21 | void09 | I get dial tone |
13:51.33 | *** join/#asterisk Dovid (~dovid@ool-4356e81f.dyn.optonline.net) |
13:51.57 | void09 | then i try to dial the number and it says "Your call cannot be completed as dialed. Please check the number and dial again" |
13:52.01 | void09 | what am I missing ? |
13:52.09 | void09 | from goip32 I can make test calls just fine |
13:52.42 | void09 | == Spawn extension (from-internal, 07, 7) exited non-zero on 'Local/07@from-internal-00000004;2' |
13:52.42 | void09 | <PROTECTED> |
13:52.42 | void09 | <PROTECTED> |
13:52.42 | void09 | <PROTECTED> |
13:53.04 | Dovid | I just joined but did you try looking at a SIP debug? Have a look at sngrep |
13:53.28 | Samot | Why are there local channels involved? |
13:53.41 | void09 | this is what it says on asterisk console |
13:53.51 | void09 | local channels? |
13:54.01 | void09 | PBX and GoIP box are on the same network |
13:54.11 | void09 | 192.168.100.198/199 |
13:54.37 | Samot | How is this call being made? |
13:56.22 | void09 | well I enter 3 (the digit to access the DISA), then enter password 1234# |
13:56.32 | void09 | then I just dial the number 07XX XXXXXX |
13:57.18 | void09 | by dialing the voip phone number in the other sip trunk, which works |
13:58.40 | void09 | https://imgur.com/a/KxRZv02 |
13:59.40 | Samot | What other SIP trunk? |
13:59.43 | void09 | IVR https://imgur.com/a/ivQio2u |
13:59.59 | void09 | the other SIP Trunk (ClickPhone) is a voip number I added. I can dial it in no problem |
14:01.02 | Samot | Show an actual sip debug of the failed call |
14:01.15 | void09 | just the text in the asterisk console ? |
14:01.20 | void09 | or where do I see this debug |
14:03.00 | void09 | https://pastebin.com/raw/6Zty3uKa |
14:04.02 | void09 | missed the first line: https://pastebin.com/raw/S9PB4tdu |
14:10.10 | void09 | just trying to call to sip clickphone, and go out through sip goip (the device with the sim card) |
14:10.34 | void09 | spent 6 hours already on this and I can't figure it out, can't find tutorial |
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14:23.19 | Samot | That's not a SIP debug |
14:23.24 | Samot | asterisk -r |
14:23.28 | Samot | sip set debug on |
14:23.38 | Samot | Make the call out the goip device |
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14:40.00 | Samot | void09: Well? |
14:58.56 | Samot | Well it's good to see 2021 is keeping the classic hits rolling. |
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15:05.47 | dongs | why is asterisk messages spamming in syslog? |
15:06.04 | dongs | in addition to /var/log/asterisk/message AND console? |
15:06.40 | Samot | What messages? |
15:07.06 | dongs | asterisk[675]: [Jan 2 00:06:38] #033[1;33mNOTICE#033[0m[804][C-00000096] and so on |
15:07.11 | dongs | including malformed ansi codes! |
15:07.19 | dongs | it looks like same contents thats going on console. |
15:07.39 | dongs | ive checked logger.conf and made syslog => error (also tried none, etc) and no effect. |
15:07.51 | dongs | also commented out syslog => completely, also same. |
15:08.25 | Samot | WAnt to show more than one line? |
15:08.32 | Samot | Give a bit more of context |
15:09.17 | dongs | http://bcas.tv/paste/results/b1gJsX49.html |
15:09.30 | dongs | i know the cause of the errors, i don't mind. |
15:09.36 | dongs | i just do NOT want them in syslog |
15:09.41 | dongs | they're already logged to asterisk messages. |
15:10.02 | Samot | What is causing the errors? |
15:10.08 | Samot | And why aren't you fixing it? |
15:10.32 | dongs | some idiots repeatedly scanning my SIP server trying to make calls from IPs that are behind nat or something and cant actually receive a SIP response back |
15:10.45 | Samot | So why don't you block them? |
15:10.55 | Samot | Instead of turning off logging that tells you bad things are happening |
15:10.59 | dongs | random IPs all the time, I dont care enough about the spam, i just dont awnt it double logged |
15:11.10 | dongs | its already logged to asterisk's own log |
15:11.15 | dongs | i don't want the spam in SYSLOG |
15:11.20 | Samot | It's not SPAM |
15:11.29 | Samot | It's the system telling you have bad things happening |
15:11.35 | Samot | Logging is not spam. |
15:11.40 | dongs | of course it is, its spam beacuse /var/log/asterisk/messages already has same stuff! |
15:11.43 | Samot | You could secure your network properly |
15:11.46 | dongs | and thats wehre I will see it if I wanted to see it |
15:11.47 | Samot | That would stop this. |
15:12.51 | Samot | What do you have in the logger config? |
15:13.05 | dongs | console => notice,warning,error |
15:13.07 | dongs | thats all. |
15:14.02 | dongs | my astlogdir points to /var/log/asterisk |
15:18.06 | void09 | Samot: just a second : ) |
15:19.45 | void09 | Samot: https://hastebin.com/ezonehatig.yaml |
15:20.53 | dongs | so.... where do i check why its double logging to syslog... |
15:21.42 | void09 | this is the output of the followinv events: dialing in the jsip voip number trunk. getting greeted by the IVR voice. pressing 3 to get to disa. entering password followed by "#". Just entering 10 digit phone number |
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15:30.19 | Samot | void09: There's no call in that debug. |
15:30.35 | void09 | well maybe that's the problem then ? :-s idk |
15:30.38 | dongs | Samot: i think i figured it out. systemd for some insane reason captures asterisk stdout and sends it to syslog when its running as a service. |
15:30.49 | void09 | the chinese "engineer" from the goip support is not online now |
15:30.52 | dongs | so this is more of a dumb ubuntu/systemd problem than asterisk |
15:31.02 | Samot | Sure. |
15:31.11 | Samot | I don't have that issue on Ubuntu. |
15:31.24 | dongs | you are probably not using their packaged version either. |
15:32.34 | Samot | Nope. I don't like being that far behind in Asterisk releases. |
15:33.19 | dongs | i just upgraded my stuff from something I setup > 10 years ago. s o i dont mind either way. my requirements are simple anyway. |
15:36.22 | dongs | got it. had to set StandardOutput=null and StandardError=null inside the unit file. i can still connect with -r so all is good. |
15:36.32 | dongs | allright, problem solved, was not asterisk issue at all. |
16:03.58 | dongs | i doubt anyone cares but I've reported it as a bug to ubuntu here https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1909816 and I do think its a bug due to double-logging and not stripping ansi codes. but yeah whatever it works so problem solved. |
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