Call failed, all lines are busy, I understand that this happens when there is no communication with the pstn but how can I see the records that help me understand the cause of the problem.
Look at your logs in /var/log/asterisk
Connect to the asterisk console ‘asterisk -r’ and see if there are any warnings or errors there.
Which channel technology are you using?
I am using elastix with sip trunk, how can I see the old trunk disconnection records? and how can I see if the network interface fell at a certain moment?
Note the Elastix GUI is not supported here. That also applies to any code in its dialplan that analyses clearing causes.
SIP or PJSIP?
For SIP, enable verbose logging and issue the CLI comman “sip set debug on” to see the actual SIP protocol exchange that is resulting in the busy status.
I called the provider of the sip trunk and they told me that it was a mistake in their platform, I wanted to check the moment when the connection between the network adapter and the sip trunk failed to identify similar problems in the future, the command " sip set debug on "allows me to see the cause of the problem or does it only work if it is in real time?
sip set debug on will allow you to see what the provider sent to you.
Only in real time? Or can you see old records?
It is logged in the verbose category, so you can see it in any logging stream enabled for that in logger.conf.
If logging was enabled and never turned off before that point of time then yes
From the console type in:
asterisk -rvvvv
sip set debug on
Place an outgoing call through ISP.
You will see SIP packets and diagnostic messages in real time.
Are you registered with the ISP?
The type sip show peers - you will see the state of your trunk with the ISP
He wants the logs in /var/log/asterisk/*, not the console output.
thank you very much I appreciate the comments, very useful