Hello!
My Flashphoner starts simultansouly phone calls through Asterisk. (a prank call project where the user can enter and call a phone number, hearing the callee and playing pre-recorded audio clips to the callee). Technically Asterisk uses MeetMe and conferences to manage the calls and Monitor for recording the phone calls. It is no problem to record the calls but only the combined, means the callee (Inbound) and the audio clips which the user starts (Outbound). But I want to seperate them. Especially I need the audio track of the callee (without the pre-recorded prank audio clips). But I am absolutely not able to manage this. I replaced Monitor with MixMonitor, also tried SpyChan, AGI scripts and else. I played around with the parameter options but always get the recordings combined. Can someone help me?
I thought that my Asterisk version on my server with WCS 2.1 is too old for using MixMonitor but then I tried it on my other server (WCS 5.2) with Asterisk 14.7.6 where all that functions are included but still I fail.
Here a short overview about my extensions.conf
Thank you in advance!
Marc
My Flashphoner starts simultansouly phone calls through Asterisk. (a prank call project where the user can enter and call a phone number, hearing the callee and playing pre-recorded audio clips to the callee). Technically Asterisk uses MeetMe and conferences to manage the calls and Monitor for recording the phone calls. It is no problem to record the calls but only the combined, means the callee (Inbound) and the audio clips which the user starts (Outbound). But I want to seperate them. Especially I need the audio track of the callee (without the pre-recorded prank audio clips). But I am absolutely not able to manage this. I replaced Monitor with MixMonitor, also tried SpyChan, AGI scripts and else. I played around with the parameter options but always get the recordings combined. Can someone help me?
I thought that my Asterisk version on my server with WCS 2.1 is too old for using MixMonitor but then I tried it on my other server (WCS 5.2) with Asterisk 14.7.6 where all that functions are included but still I fail.
Here a short overview about my extensions.conf
Code:
[general]
static = yes
writeprotect = yes
autofallthrough = yes
priorityjumping = yes
clearglobalvars = yes
[globals]
CONSOLE=Console/dsp
;# IAXINFO=guest
exten => _0049ZX.,1,NoOp(Starting Dialout Procedure app_sound_voippro INT ${EXTEN})
same => n,SIPAddHeader(P-Preferred-Identity: <sip:+49xxxxxxx@sip.telephone.org\;user=phone>)
same => n,Set(CALLERID(all)=+49xxxxxxxxx)
same => n,Set(__CONF=${EXTEN})
same => n,Set(__INIT_CHANNEL=${CDR(channel)})
;same => n,AGI(check_channel.sh,${INIT_CHANNEL}xxx${CONF})
same => n,Set(FILE(/var/lib/asterisk/conferences/${CONF})=${SIPCALLID})
same => n,Set(MONITOR_FILENAME=/srv/www/marcophono/html/recs/${EXTEN:1})
same => n,Monitor(wav,${MONITOR_FILENAME},i)
same => n,Dial(SIP/${EXTEN}@sipout_teleflash,30,rG(separator,s,1))
same => n,NoOp("Call should not end here")
same => n,Hangup()
[separator]
exten => s,1,Goto(fun1,s,1)
exten => s,2,Goto(fun2,s,1)
[fun1]
exten => s,1,NoOp(Data 1: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,Set(TIMEOUT(absolute)=305)
exten => s,n,MeetMe(${CONF},Akmqd)
exten => s,n,Hangup
exten => h,1,AGI(kill_channel.sh,${INIT_CHANNEL})
[fun2]
exten => s,1,NoOp(Data 2: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,Wait(0.5)
exten => s,n,Set(TIMEOUT(absolute)=305)
exten => s,n,MeetMe(${CONF},kxqd)
exten => s,n,AGI(kill_channel.sh,${INIT_CHANNEL})
exten => s,n,NoOp("NOT_IN_THE_MEETME")
exten => s,n,Hangup
exten => h,1,AGI(kill_channel.sh,${INIT_CHANNEL})
[inject-sound]
exten => s,1,MeetMe(${CONF},qd)
exten => s,n,NoOp("AFTER MEETME")Marc
