root@sysadm:/etc/asterisk# cat sip.conf

root@sysadm:/etc/asterisk# cat extensions.conf

 
* [general]
* static=yes
* writeprotect=yes
* clearglobalvars=no
* 
* [globals]
* 
* 
* [from-net]
* 
* exten => _XXXX,1,Dial(SIP/${EXTEN},10)
* same => n,Hangup()
* 
* exten => _XXXX/_1003,1,NoOp(-----)
* exten => _XXXX/_1003,2,Set(CALLERID(NAME)=Helpdesk)
* exten => _XXXX/_1003,3,Set(CALLERID(NUM)=09666666666)
* exten => _XXXX/_09666666666,4,Dial(SIP/${EXTEN},10)
* same => n,Hangup()
* 
* 
* 
* ;exten => _0X./_1001,1,Dial(SIP/BDCOM/${EXTEN})
* ;same => n,Hanup()
* 
* ;exten => 1002,1,Dial(SIP/1002)
* ;same => n,Hnaup()
* 
* ;exten => 1001,1,Dial(SIP/1001)
* ;same => n,Hnaup()
* 
* ;exten => 1009,1,Dial(SIP/1009)
* ;same => n,Hnaup()
* 
* [from-bdcom]
* 
* ;exten => _911322,1,Dial(SIP/1001)
* ;same => n,Hangup()
* 
* exten => _911322,1,Read(INPUT,/home/sysadm/bdcomMain)
* same => n,NoOp(---${INPUT}---)
* same => n,Goto(from-input,${INPUT},1)
* ;same => n,Dial(SIP/${INPUT},10)
* ;same => n,Hangup()
* 
* [from-input]
* exten => _0,1,Queue(support)
* same => n,Hangup()
* 
* exten => _1,1,MixMonitor(/home/sysadm/recording/${UNIQUEID}.wav,b)
* same => n,GotoIfTime(06:30-17:30,*,*,*?open:close)
* 
* same => n(open),Dial(SIP/1001)
* same => n,Hangup()
* 
* same => n(close),Dial(SIP/1009)
* same => n,Hangup()
* 
* 

root@sysadm:/etc/asterisk# cat cdr_mysql.conf

http://downloads.asterisk.org/pub/telephony/asterisk/

https://www.voip-info.org/asterisk-config-queuesconf /?cf_chl_jschl_tk=pmd_Nrl.DSBxcF9YnFbDRweEw7EUmFA4yejMQSfvAA0W260-1635147369-0-gqNtZGzNAnujcnBszQf9

https://wiki.asterisk.org/wiki/display/AST/Building+Queues

https://wiki.asterisk.org/wiki/display/AST/The+Read+Application

Asterisk CLI

Type “asterisk -r” to bring you into the command line for your PBX

Type “sip show registry” to show you your trunk registrations

Type “sip show peers” to show your extensions

Type “sip show peer 100” to show you details of that extension (note: use the extension number you created if it’s not 100)

Type “sip set debug on” to enable debugging and watch the SIP traffic

Using the above commands, and monitoring information on a busy phone system can be difficult due to the speed of the information that is coming in on your screen. The commands listed below would be best suited for busier environments as they remove a lot of noise that can occur while watching the SIP traffic.

Type “exit” and hit enter to get back to the main command line prompt for your PBX

Type “cd /var/log/asterisk” to view all of the logs

Type “tail full” to view the full logs only

Type “asterisk -r” to go back and write the command to remove debugging

Type “sip set debug off” to turn off debugging, to avoid information overload when viewing your logs

Type “exit” and hit enter to go back to the main command line prompt for your PBX

Type “more full : grep 100” to give you all of the messages that include “100” (if different, replace 100 with the extension number that you created)

Type “more full : grep -i unreachable” to view logs that indicate the extension was unreachable

sip show peer 183

Description :

Secret       : <Set>
MD5Secret    : <Not set>
Remote Secret: <Not set>
Context      : from-vip
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language     : 
Tonezone     : <Not set>
AMA flags    : Unknown
Transfer mode: open
CallingPres  : Presentation Allowed, Not Screened
Callgroup    : 1
Pickupgroup  : 1
Named Callgr : 
Nam. Pickupgr: 
MOH Suggest  : 
Mailbox      : 183@default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit   : 2
Max forwards : 0
Dynamic      : Yes
Callerid     : "Md. Ziaul Haque Chowdhury Raju" <>
MaxCallBR    : 384 kbps
Expire       : 3488
Insecure     : port,invite
Force rport  : Yes
Symmetric RTP: Yes
ACL          : Yes
ContactACL   : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia  : No
PromiscRedir : No
User=Phone   : No
Video Support: Yes
Text Support : No
Ign SDP ver  : No
Trust RPID   : Yes
Send RPID    : Yes
Path support : No
Path         : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode     : rfc2833
Timer T1     : 500
Timer B      : 32000
ToHost       : 
Addr->IP     : 192.168.1.66:5060
Defaddr->IP  : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 183
SIP Options  : (none)
Codecs       : (alaw|ulaw)
Auto-Framing : No
Status       : OK (10 ms)
Useragent    : Grandstream GXP1625 1.0.4.132
Reg. Contact : sip:183@192.168.1.66:5060
Qualify Freq : 60000 ms
Keepalive    : 0 ms
Sess-Timers  : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess     : 90 secs
RTP Engine   : asterisk
Parkinglot   : 
Use Reason   : No
Encryption   : No
RTCP Mux     : No

https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients

https://www.voip-info.org/asterisk-cli/?__cf_chl_jschl_tk__=pmd_89b813965b0cd6f1dab848a04629eda659048886-1629098559-0-gqNtZGzNAiKjcnBszQ0O