Free IVR and voicemail system for MS Lync/SFB Part 2

Welcome Back. In Part 1 we finished installing asterisk so let start configuring everything.

Asterisk Basic files

• first you need to configure some basic configuration, you have two options to do so:
• simple one by just run "make samples" but this will enable a lot of features you may don't need to use them
• just create two essential files called (asterisk.conf, and modules.conf) and put them in "/etc/asterisk" folder (press on file names to see sample of confiugration).
• run "service asterisk restart" to reset the asterisk service to update configuration.

Asterisk SIP Trunks configuration

• The sip configurations are located in file called "sip.conf" at "/etc/asterisk" so create the file and open it.
• first you need to enable SIP/TCP as a transport as Lync/SFB work only with TCP and asterisk by default support UDP, you could do it by write below configuration, 

[general]

tcpenable=yes

tcpbindaddr=0.0.0.0

transport=tcp,udp

• We will need to have two sip trunks one with Lync/SFB and one with the gateway on same file add below lines

[Lync]

disallow=all

host=<hostname of Lync/SFB FE server>

type=peer

insecure=port,invite

transport=tcp

port=<TCP Port of Lync/SFB mediation>

context=Lync-Trunk

dtmfmode=RFC2833

allow=alaw

allow=ulaw

[Gateway]

disallow=all

host=<hostname of gateway>

type=peer

insecure=port,invite

transport=tcp "you could use UDP of gateway if it doesn't support TCP"

port=<port of Gateway>

context=Gateway-Trunk

dtmfmode=RFC2833

allow=alaw

allow=ulaw

• run "service asterisk restart" to reset the asterisk service to update configuration.

Asterisk IVR sound files

• First you need to import sound files to asterisk these file should be in .WAV format.
• According to my workflow from part1 we will need these files "Welcome, Main_Ar, Transfer_Ar, Wrong_Ar, Main_En, Transfer_En, Wrong_En)".
• Sure you could have you own files and also IVR workflow this is just a sample.
• The files should have below characteristics to work with
• If you record in a studio, use 8kHz 16 bits, not 8 bits, and phone line is 12 to 13 bits compressed down to 8 in a pseudo-logarithmic way
• then down-sample your audio to 8000hz and save in Windows PCM wav (16 bit) format
• When recording find a really quiet place. Background noise is usually the biggest hassle when recording prompts
• use a reasonable mic; fix it down someplace (don't hand hold it) 
• Upload these files to "/var/lib/asterisk/sounds/en/"
• Then we need to convert them to format that asterisk understand (gsm, or sln) so you should run below command for every file."sox /var/lib/asterisk/sounds/en/Welcome.wav -t raw -r 8000 -s -2 -c 1 /var/lib/asterisk/sounds/en/Welcome.sln"

Asterisk IVR configuration

• you need to create one of the most important files in asterisk which called "extensions.conf" it is also exist in "/etc/asterisk"
• first you need to define a number that the Gateway will send all incoming call to it in my configuration below it is "299"
• According to my workflow from part1 i have written below in the file, sure you need to change it if your workflow is different.

; trunk assign to call received from gateway
 [Gateway-Trunk]

; answer the received call on 299 number

exten => 299,1,answer()

; play welcome and wait maximum 3 digit

  same => n,Read(extens,Welcome,3,,,5)

; if the caller didn't give any digits go to arabic IVR_AR

  same => n,gotoIf($[${LEN(${extens})} == 0]?IVR_AR,299,Start) 

; if the caller enter 2 then go to english IVR_EN

  same => n,gotoIf($[${extens} == 2]?IVR_EN,299,Start)

; if the user out 3 digit go to direct to call internal extension

  same => n,gotoIf($[$[${LEN(${extens})} = 3]]?IVR_AR,299,Transfer)

; if non of the above run the message again

  same => 299,6,goto(,299,1)

;arabic IVR section

[IVR_AR] 

; play main and wait for 3 digits

exten => 299,1(Start),Read(extens,Main_Ar,3,,,2)

;if the user out 3 digit go to direct to call internal extension

  same => n,GotoIf($[$[${LEN(${extens})} = 3]]?:IVR_AR,299,Start)

; this is the section of direct call of internal ext 

; play transfer msg before dial

  same => n(Transfer),Playback(Transfer_Ar) 

; dial internal ext at lync

  same => n,Dial(SIP/${extens}@Lync) 

; if call answered go to end 

  same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?stop) 

 ; if user doesn't exist go to wrong ext

  same => n,GotoIF($["${DIALSTATUS}" = "CHANUNAVAIL"]?wrong) 

; if non of above it mean missed call go to voicemail of this extension

  same => n,VoiceMail(${extens}@default) 

; end call

  same => n,Hangup() 

; this is wrong extension section

; play wrong ext msg

  same => n(wrong),Playback(Wrong_Ar)

; return back to start of arabic IVR_AR

  same => n,Goto(,299,Start) 

; end call

  same => n(stop),Hangup() 

;enlgish IVR same like arabic but with english msgs

[IVR_EN]

exten => 299,1(Start),Read(extens,Main_En,3,,,2)

  same => n,GotoIf($[$[${LEN(${extens})} = 3]|$[${extens} == 0]]?:IVR_EN,299,Start)

  same => n(Transfer),Playback(Transfer_En)

  same => n,Dial(SIP/${extens}@Lync_Out)

  same => n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?stop)

  same => n,GotoIF($["${DIALSTATUS}" = "CHANUNAVAIL"]?wrong)

  same => n,VoiceMail(${extens}@default)

  same => n,Hangup()

  same => n(wrong),Playback(Wrong_En)

  same => n,Goto(,299,Start)

  same => n(stop),Hangup

check  Part 1 and Part 3

See you in part 3