Clear log file access

Hi everyone, I ran the command sudo asterisk -rvvvv and there are many failed login attempts. I would like to know how to clean this list in order to gradually check for each attempt what the problem may be. So I would like to clean up this list how can I do?

These are in a log. Individual log entries are never removed by Asterisk as they are statements of historic truth.

Also, exactly what is the message, and are you using a GUI?

i am trying to log in via linphone app from my mobile. asterisk sees the attempts but does not connect me

I assume you mean by log in that you are trying to register a SIP user agent.

What appears on the logs, at at least verbosity 5?

What does “psip set logger on” show as being received from and sent to the device. (If you are not using chan_pjsip, you should change to that first.)

What do you have as your configuration for the device, and for the transport, in pjsip.conf?

I saw the pjsib file. I downloaded the latest version of asterisk. But I changed the sip.conf file not pjsip.conf; I bought a basic course on udemy it seems that everyone can do the exercise but I can’t.

When I try to connect with my mobile, it is connected. when i log into the asterisk cli and try to make a call from another user, i get this error:

[Apr 29 12:38:38] NOTICE [1768] [C-00000005]: chan_sip.c: 26824 handle_request_invite: Call from ‘5100’ (93.32.165.175:47616) to extension ‘5001’ rejected because extension not found in context 'myphones

call has been declined appears on my mobile phone
I changed extensions for 5101 instead of 5001 but it falls. I changed it because previously it gave me another error.

Current situation of the sip.conf and extensions files place only what I have changed:

sip:

[5100]
type=friend
secret=5100
context=myphones
host=dynamic

[5101]
type=friend
secret=5101
context=myphones
host=dynamic

[5102]
type=friend
secret=5102
context=myphones
host=dynamic

[general]
context=public ; Default context for incoming calls. Defaults to ‘default’
nat=force_rport,comedia
localnet=172.31.18.78/255.255.240.0
externip=52.203.197.19
canreinvite=no
videosupport=yes
dtmfmode=rfc2833
qualify=yes
disallow=all
;codec for audio
allow=gsm
;codec for video
allow=vp8
allow=h264
nat=yes

extensions.conf:

[myphones]
exten => 5100,1,Dial (SIP/5100)
exten => 5100,n,Hangup

exten => 5101,1,Dial (SIP/5101)
exten => 5101,n,Hangup

exten => 5102,1,Dial (SIP/5102)
exten => 5102,n,Hangup

If you are on a course that both tells you to use chan_sip, and also to use obsolete parameter names and set unnecessary options (twice!), I’d find a better course.

The message is correct and clear for your current configuration.

I’d suggest that any course based on chan_sip is at least six years out of date, and any course suggesting the use of canreinvite hasn’t been properly reviewed for at least 10 years.

You shouldn’t be running courses based on a deprecated feature that is going to be removed in next year’s version.

I tried to see something but there are no official courses and the book made available asterisk 4 edition is not like a sybex book which are self-learning books but simply guides. In practice there are no official asterisk courses as they do for linux lpi or cisco. The version of asterisk that I use in the course that I bought is asterisk 17.6, while I have installed the latest version of asterisk 18.0

There are official courses, although I’m not in a position to say if they are any good.

My account is not set up to have permissions to access the courses maybe they are not for everyone

in your [myphones] sektion you have have 5100 / 5101 / 5102
you need dial one of those numbers

I dialed these numbers I set these numbers in the extensions and sip file. From the application I then simply made a call to one of these numbers

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