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    Wednesday, March 06, 2013

    Advanced Configuration of Asterisk CLI

    In my last post I had shown you some very basic configuration steps of Asterisk. Now I will show you Asterisk configuration including pattern matching context.

    Configuring SIP accounts

    # cd /etc/asterisk
    # vi sip.conf [1000] deny= secret=1000 dtmfmode=rfc2833 canreinvite=no context=intercalling host=dynamic type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=udp encryption=no callgroup= pickupgroup= dial=SIP/1000 permit= [1001] deny= secret=1001 dtmfmode=rfc2833 canreinvite=no context=intercalling host=dynamic trustrpid=yes sendrpid=no type=friend nat=no port=5060 qualify=no qualifyfreq=60 transport=udp encryption=no callgroup= pickupgroup= dial=SIP/1001 permit=
    type — type of connection (peer — outcoming calls only, user — incoming calls, friend — both incoming and outcoming calls)
    context — context in /etc/asterisk/extension.conf that will be applied when a number is dialed
    secret — password for a SIP phone
    host — phone host name (dynamic host name)
    nat — network address translation
    qualify — setting it to YES will automaticly send OPTIONS packet after every 2000msec to the endpoint
    canreinvite — reinvite policy for this device
    callgroup — call group number where this device is a part of
    pickupgroup — This device can pickup calls from any group. Device doesnot have to be in any group to pickup calls
    dtmfmode=auto — dtmf mode (auto|inband|info|rfc2833)
    disallow=all — disallows all codecs
    allow=g722 — allows using codec g722
    deny - IP address range where the access has to be denied
    permit - IP address range to allow access from the client machine

    Configuring Dial plan extensions

    # vi extensions.conf
    include => intercalling
    ; If nobody picks up within 30 seconds, the call is sent to  voicemail
    ; If the extension is busy, the call is sent to voicemail
    exten => _100X,1,Set(TARGETNO=${EXTEN})
    exten => _100X,n,Dial(SIP/${EXTEN},30)
    ; routes the call to the status priority (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => _100X,n,Goto(s-${DIALSTATUS},1)
    ; Person at extension "is unavailable" message
    exten => s-NOANSWER,1,VoiceMail(${TARGETNO},u)
    ; Person at extension "is busy" message
    exten => s-BUSY,1,VoiceMail(${TARGETNO},b)
    ; To be safe, clean up the call after an answer by hanging up exten => s-ANSWER,1,Hangup()
    ; Handle any unhandled status the same way we handle NOANSWER exten => _s-.,1,Goto(s-NOANSWER,1)
    For detailed information on all the used syntax above please follow this link
    The Asterisk Book

    Let me know your success and failures.

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    Connecting two IP PBX box using SIP Trunk
    Hanging up active calls in Asterisk PBX

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