commanderk33n / BSD-Asterisk

create a asterisk VoIP-Server in FreeBSD

BSD-Asterisk

create a asterisk VoIP-Server in FreeBSD

1. Install Asterisk
   pkg install asterisk13
   to run server automatically on startup add the following line into /etc/rc.conf:
   asterisk_enable="YES"
   start the server:
   /usr/local/etc/rc.d/asterisk start
   check if server is running:
   ps ax | grep asterisk
   

2. Initial Configuration
   location of the config-files:
   /usr/local/etc/asterisk/
   basic sip.conf example:

   [general]
   context=unauthenticated ; default context for incoming calls
   allowguest=no ; disable unauthenticated calls
   srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
   udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
   tcpenable=no ; disable TCP support
   
   [office-phone](!) ; create a template for our devices
   type=friend ; the channel driver will match on username first, IP second
   context=LocalSets ; this is where calls from the device will enter the dialplan
   host=dynamic ; the device will register with asterisk
   nat=yes ; assume device is behind NAT
   secret=123; a secure password for this device -- DON'T USE THIS PASSWORD!
   dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
   disallow=all ; reset which voice codecs this device will accept or offer
   allow=ulaw ; which audio codecs to accept from, and request to, the device
   allow=alaw ; in the order we prefer
   allow=all
   textsupport=yes
   [alice](office-phone)
   [bob](office-phone) 
   

   basic extension.conf example:

   [general]
   [LocalSets]
   exten => 99,1,VoiceMailMain()
   exten = 1337,1,Answer()
   same = n,Wait(1)
   same = n,Playback(hello-world)
   same = n,Hangup()
   
   exten = 2000, 1, Dial(SIP/alice)
   exten = 2001, 1, Dial(SIP/bob)
   

3. monitoring&controlling
   enter asterisk CLI with the following command:
   asterisk -r
   show all sip user:
   sip show peers
   show all call connections:
   core show channels
   find useful commands:
   core show help
   get info how to use a command:
   core show help sip show channel
   

   reload configuration files:

   sip reload     # reload sip.conf 
   iax2 reload    # reload iax.conf 
   core reload    # reload entire configuration
   

   stop server:
   core stop now
   getting useful informations:

   sip show channel     # Show detailed SIP channel info
   sip show domains     # List our local SIP domains
   sip show history     # Show SIP dialog history
   

4. debuggin and logs
   logfile location can be configured in asterisk.conf
   edit logger.conf to enalbe specific debug output to your filesystem:
   

   [logfiles]
   debug_log_123456 => notice,warning,error,debug,verbose,dtmf
   

   enable/disable live debug in CLI:

   sip set debug on/off
   iax2 set debug on/off
   

5. iax.conf
   Server A iax.conf example:

   [general]
   [ast2]
   host=192.168.0.XX
   type=friend
   trunk=yes
   qualify=yes
   secret=1234
   context=LocalSets
   

   Server B iax.conf example:

   [general]
   [ast1]
   host=192.168.0.XX
   type=friend
   trunk=yes
   qualify=yes
   secret=1234
   context=LocalSets
   

   Dialplan example:

   exten => 09021001,1,Dial(IAX2/ast2/1001)
   ;          ^   ^               ^     ^
   ;          |   |               |     |
   ; virt.prefix ext      connection   ext
   

   SIP-User with the number 1001 on Server B is callable with the number 09021001