Hello, I will try to answer the split questions.
1- In Asterisk software, it is possible to debug a phone number only? without it being peer from my server?
You can filter through the IP of this peer, as you already know, but if this is not enough you can use wireshark and filter the captures by voip calls, by source ip, destination, package content (filtering by peer, for example) etc.
Reading suggestion: Wireshark - Voip Calls
2- Similarly to debug it is possible to control the verbose for only one user?
Yeah, but by the log instead of the CLI. You can filter calls to only one user through the Asterisk log, as long as the verbose log is enabled in the /etc/Asterisk/logger.conf
Example:
full => notice,warning,error,verbose,dtmf,fax
From Asterisk 11 every call has an identifier (call-id) which makes it possible to track it in the log. Example:
[2017-04-11 09:17:02] VERBOSE[50999][C-00000620] pbx.c: Executing [1234@CONTEXTO-PADRAO:1] NoOp("SIP/PEERQUALQUER-00000507", "### RECEBENDO CHAMADA DE 1234567890 PARA 1234 ###") in new stack
On the above line you can see that Asterisk received a call from the PEERQUALQUER peer to the 1234 extension. All lines of this call have the [C-00000620] identifier that can be filtered with a 'grep', for example:
grep "C-00000620" /var/log/asterisk/full
Reading suggestion: Call Identifier Logging
3- If it is not possible there is some standardized way to make tests of this type in production? to filter calls incomming
I use the log file data to filter and analyze incoming calls. When I want more details, mainly of signaling, I use the sngrep for real-time analysis of SIP messages, dialogues and transactions. I use wireshark for pcap file analysis when I don’t need real-time analysis.