• VMware

    Learn about VMware virtualization for its products like vsphere ESX and ESXi, vCenter Server, VMware View, VMware P2V and many more

  • Linux

    Step by step configuration tutorials for many of the Linux services like DNS, DHCP, FTP, Samba4 etc including many tips and tricks in Red Hat Linux.

  • Database

    Learn installation and configuration of databases like Oracle, My SQL, Postgresql, etc including many other related tutorials in Linux.

  • 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=0.0.0.0/0.0.0.0 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=0.0.0.0/0.0.0.0 [1001] deny=0.0.0.0/0.0.0.0 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=0.0.0.0/0.0.0.0
    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
    [general]
    static=yes
    writeprotect=no
    clearglobalvars=no
    
    [default]
    include => intercalling
    
     [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.

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


    Deepak Prasad

    is a techie and an author who is still trying to survive in this IT generation with very little knowledge he has on Linux/Unix, VMware, SAN Storage, Automation, networking etc

    You can follow him on Facebook or Google+


    Do you also have something to share here?
    Join GoLinuxHub Team as an Author, Click here for more information
    Advanced Configuration of Asterisk CLI Advanced Configuration of Asterisk CLI Reviewed by Deepak Prasad on Wednesday, March 06, 2013 Rating: 5

    No comments:

    Powered by Blogger.