VoIP: Asterisk pakai GotoIf

Sumber: http://www.spinics.net/lists/asterisk/msg157160.html

exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr)
exten => _7NXXXXXX,n,NoOp(DialStatus="${DIALSTATUS}")
exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2)
exten => _7NXXXXXX,n,Hangup()
exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr)
exten => _7NXXXXXX,n,NoOp(DialStatus="${DIALSTATUS}")
exten => _7NXXXXXX,n,Hangup()

atau

exten => _08.,1,Dial(SIP/${EXTEN}@gw-ext1,60,tr)
exten => _08.,n,NoOp(DialStatus="${DIALSTATUS}")
exten => _08.,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2)
exten => _08.,n,Hangup()
exten => _08.,n(line2),Dial(SIP/${EXTEN}@gw-ext2,60,tr)
exten => _08.,n,NoOp(DialStatus="${DIALSTATUS}")
exten => _08.,n,Hangup()

Beberapa hasil dari DIALSTATUS adalah

• ANSWER: Call is answered. A successful dial. The caller reached the callee.
• BUSY: Busy signal. The dial command reached its number but the number is busy.
• NOANSWER: No answer. The dial command reached its number, the number rang for too long, then the dial timed out.
• CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up.
• CONGESTION: Congestion. This status is usually a sign that the dialled number is not recognised.
• CHANUNAVAIL: Channel unavailable. On SIP, peer may not be registered.
• DONTCALL: Privacy mode, callee rejected the call
• TORTURE: Privacy mode, callee chose to send caller to torture menu
• INVALIDARGS: Error parsing Dial command arguments (added for Asterisk 1.4.1, SVN r53135-53136)

Lebih Dalam

• VoIP: Trunk
• VoIP: Asterisk menerima Anonymous Call
• VoIP: Asterisk route ke arah IP PBX lain dengan kode area
• VoIP: Cara Mengkonfigurasi Trunk di Asterisk
• VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk
• VoIP: Asterisk forward call ke IP softswitch lain

Asterisk Round Robin

• VoIP: Astersk Dial Round Robin
• VoIP: Asterisk pakai GotoIf

OpenSIPS

• VoIP: OpenSIPS route ke arah Asterisk
• OpenSIPS: Rewrite URI
• OpenSIPS: Rewritehostport

OpenSIPS round robin

• OpenSIPS: dispatcher

Referensi

• http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS