Chapter 3. Installing and Configuring Asterisk

If you want to configure and make Asterisk you will need to install dahdi-linux first, then dahdi-tools. Dahdi is needed to run telephony conference.

 

Installing Dahdi-linux

 The required version of dahdi-linux is 2.0.0, which you can download here:

http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.co...

Do the following:

  • Copy Dahdi-linux files to /usr/src/dahdi-linux
  • In the directory /usr/src/dahdi-linux/ type make in the command line. Press Enter.
  • Type make install in the command line. Press Enter.

  

Installing Dahdi-tools

 The required version of dahdi-tools is 2.0.0, which you can download here:

http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.co...

Do the following:

  • Copy Dahdi-tools files to /usr/src/dahdi-tools
  • Go to the /usr/src/dahdi-tools/ directory and type ./configure in the command line. Press Enter
  • Type make in the command line. Press Enter.
  • Type make install in the command line. Press Enter.

  

Installing Asterisk

 The required version of Asterisk is 1.6.0.1, which you can download here:

http://www.asterisk.org/

Do the following:

  • Copy Asterisk files to /usr/src/asterisk.
  • In configuration file /usr/src/asterisk/include/asterisk/channel.h, set the value AST MAX EXTENSION equal to 256.
  • Go to /usr/src/asterisk/ directory and type ./configure in the command line. Press Enter
  • To initiate the building process type make in the command line. Press Enter.
  • After building is completed, type make install. Press Enter.
  • Type make samples in the command line. Press Enter.

  

Installing AMD (Answering Machine Detector)

 AMD is an optional Asterisk application, enabling the system to detect answering machines on response.

Do the following:

  • Download to /etc/asterisk amd.conf
  • Execute 'dos2unix amd.conf'
  • Execute 'vi amd.conf' and change [AnsweringMachineDetector] to [amd]
  • Download to /usr/src/asterisk/apps app_amd2.c
  • Execute 'mv app_amd2.c app_amd.c'
  • Execute 'dos2unix app_amd.c'
  • In your /usr/src/asterisk/apps directory, edit Makefile by adding asterisk modul “app_amd.so” after the list of other moduls which are after “APPS=”
  • Go back to your /usr/src/asterisk directory. Execute “make” and then “make install”.

  

Adding Configuration Files

 Do the following:

  • Copy extensions.conf, iax.conf and sip.conf files to your /etc/asterisk directory.
  • Execute ‘dos2unix’ to each one.

 Now you are ready to configure these files with respect to your personal settings.

 

Editing Configuration Files

 There are several files copied to your /etc/asterisk directory. The following describes how to configure each one.

  • amd.conf

 In this file nothing has to be changed. This file describes parameters needed for answering machine detection.

  • sip.conf

The first section of the configuration file describes general SIP configuration parameters. Nothing needs to be changed there.

    [general]
    context=default
    allowguest=yes
    bindport=5060
    bindaddr=0.0.0.0
    srvlookup=yes
    disallow=all
    allow=ulaw
    allow=alaw
    promiscredir=yes
    canreinvite=no
    dtmfmode=rfc2833
    nat=yes
    musiconhold=none
    ; Mininum roundtrip time for messages to monitored hosts
    t1min=1000

    The second section of the configuration file describes parameters for DID-gateway. Sometimes, a SIP-provider requires registration; that is why there is a long direct DID-number (e.g. 6786081290 – in the template file).


    ; format: register => user[:secret[:authuser]]@host[:port][/extension]
    ; sample
    register => 6786081290@sip-truck.starpoundtech.net:9000/6786081290

    The next section contains parameters needed for outgoing calls. Required information is username, host and port. Please note that this section is needed only if Asterisk server is used for SIP trunking to PSTN.

    [sip-outbound]
    type=peer
    username=[username]
    host=[hostname]
    port=[port]
    fromdomain=[domain_name]
    canreinvite=no
    insecure=very
    qualify=no
    nat=yes
    context=pstn-in

    The last outline is needed for the loop connection of the Border Controller (Asterisk) and SER. In this outline, you need to set your IP instead of string.


    [trunk-ser]
    type=peer
    host=[Kamailio_IP] nat=yes
    canreinvite=nov

    • extensions.conf
    Border Control configuration. This is specified within the Redirect Calls outline. Nothing has to be changed in this outline.

    ; configuration for using Asterisk as a SIP Server
    ; redirect calls back to Kamailio
    exten => _frm_ser.,1,Set(from=${SIP_HEADER(From)})
    exten => _frm_ser.,2,SIPAddHeader(SPFrom: ${from})
    exten => _frm_ser.,3,Set(to=${SIP_HEADER(To)})
    exten => _frm_ser.,4,SIPAddHeader(SPTo: ${to})
    exten => _frm_ser.,5,Set(callId=${SIP_HEADER(Call-ID)})
    exten => _frm_ser.,6,SIPAddHeader(SPCallId: ${callId})
    exten => _frm_ser.,7,Dial(SIP/frm_sbc${EXTEN:7}@trunk-ser,60,r)
    exten => _frm_ser.,108,Hangup()
    exten => _frm_ser.,8,Hangup()

    ; launching media queues
    ; format: ${EXTEN}@[Application_server_IP]:[Application_server_SIP_port]
    exten => lmtt,1,Dial(SIP/${EXTEN}@172.26.10.39)
    exten => lmtt,102,Hangup()

    DID Gateway configuration. This is specified within the Outbound Calls outline. Set the peer name the same as in sip.conf.

     


    ; outbound calls
    ; sample
    exten => _9.,1,Set(LIMIT_PLAYAUDIO_CALLEE=yes
    ) exten => _9.,2,Dial(SIP/${EXTEN:1}@voice-ring-6786081290,60,rL(180000:20000))
    exten => _9.,3,Hangup()
    exten => _9.,103,Hangup()

    AMD configuration. This is specified within the Outbound Calls with Answering machine detection outline. Nothing has to be changed there.


    ; outbound calls with Answering Machine Detection
    exten => _amd.,1,Set(LIMIT_PLAYAUDIO_CALLEE=yes)
    exten => _amd.,2,Dial(SIP/${EXTEN:3}@voice-ring-6786081290,60,rM(amd)L(180000:20000))
    exten => _amd.,3,Hangup()
    exten => _amd.,103,Hangup()

    Media Server configuration. While using StarPound 1.1.0 RC2 ., you might need more than one media queue. This is set in the Processing Media Queue outline. Use the default queue as a template: character “_” specifies a mask, character “X” means “any digit 0-9”.
    Note: after changing any configuration files, you need to restart Asterisk to proceed.

    ; processing media queues
    exten => _pmq.,1,Answer
    exten => _pmq.,n,SetMusicOnHold(default)
    exten => _pmq.,n,MusicOnHold()

    ; playback media file
    exten => playmedia,1,Answer
    exten => playmedia,n,Set(MUSICFILE=${SIP_HEADER(SPMusicFile)})
    exten => playmedia,n,Playback(${MUSICFILE})
    exten => playmedia,n,Wait(999)

    ; enter conference
    exten => conference,1,Answer
    exten => conference,n,Set(CONFID=${SIP_HEADER(SPConfIdHeader)})
    exten => conference,n,Meetme(${CONFID}|qd)

    ; enter conference (listen only mode)
    exten => listenIn,1,Answer
    exten => listenIn,n,Set(CONFID=${SIP_HEADER(SPConfIdHeader)})
    exten => listenIn,n,Meetme(${CONFID}|qdm)

    ; music on hold extension
    exten => hold,1,Answer
    exten => hold,n,SetMusicOnHold(default)
    exten => hold,n,MusicOnHold()

    ; ringing extension
    exten => ringing,1,Answer
    exten => ringing,n,Playtones(ring)
    exten => ringing,n,Wait(${MAX_TIMEOUT})
    exten => ringing,n,HangUp()

    ; dtmf extension
    exten => dtmf,1,Answer
    exten => dtmf,n,Set(digits=${SIP_HEADER(SPDigitsHeader)})
    exten => dtmf,n,SIPDtmfMode(rfc2833)
    exten => dtmf,n,SendDTMF(${digits})
    exten => dtmf,n,Hangup()

    ; whisper
    exten => whisper,1,Answer
    exten => whisper,2,Set(filename=${SIP_HEADER(SPFilenameHeader)})
    exten => whisper,3,Playback(${filename},j)
    exten => whisper,4,Hangup()
    exten => whisper,104,Wait(1)
    exten => whisper,105,Hangup()

    ; record
    exten => record,1,Answer
    exten => record,n,Set(filename=${SIP_HEADER(SPFilenameHeader)})
    exten => record,n,Set(callBackNumber=${SIP_HEADER(SPCallBackHeader)})
    exten => record,n,Monitor(wav,/var/lib/asterisk/sounds/recorded/${filename},m)
    exten => record,n,Dial(${callBackNumber})
    exten => record,n,Hangup()

    ; simple ivr extension
    exten => ivr,1,Answer
    exten => ivr,n,Set(ip=${SIP_HEADER(SPAppServerIp)})
    exten => ivr,n,Set(port=${SIP_HEADER(SPAgiPort)})
    exten => ivr,n,Set(callId=${SIP_HEADER(SPCallId)})
    exten => ivr,n,AGI(agi://${ip}:${port}/hello.agi?callId=${callId})

    Click2Talk configuration. This is specified in the last configuration outline. Nothing has to be changed there; however, you need to set some parameters in iax.conf.

    ; click2talk configuartion

    ; click2talk extension, format: collab?[Launching_media_queue]?[Application_server_IP]?[Application_server_port]?[Custom_params]
    exten => _collab?.?.?.,1,Set(launch=${CUT(EXTEN,?,2)})
    exten => _collab?.?.?.,n,Set(ip=${CUT(EXTEN,?,3)})
    exten => _collab?.?.?.,n,Set(port=${CUT(EXTEN,?,4)})
    exten => _collab?.?.?.,n,Set(params=${CUT(EXTEN,?,5)})
    exten => _collab?.?.?.,n,SIPAddHeader(SPParameters: ${params})
    exten => _collab?.?.?.,n,Dial(SIP/${launch}@${ip}:${port})

    AMD macros. Nothing has to be changed there.

    [macro-amd]
    exten => s,1,AMD
    exten => s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
    ;exten => s,n(mach),WaitForSilence(500)
    exten => s,n(mach),SetVar(MACRO_RESULT=ABORT)
    ;exten => s,n(humn),WaitForSilence(500)
    exten => s,n(humn),NoOp

    • iax.conf

     

This file configures Click2Talk feature. You will need to set your IP and port within General IAX Configuration outline. You can create users in IAX Users outline, using the default configuration as a template.