Hello everyone,
I am now using Asterisk 16.19.1 since a month and I am very happy to got it working.
I am looking to install instant messaging in MicroSIP.
Currently when I send a message from 1067 to 1092, the 1092’s MicroSIP is ringing as anonymous with no sound.
Here is my extension.conf :
[Aucun] ; No plan
exten => _.,1,PlayBack(/var/lib/asterisk/sounds/custom/noforfait)
[get-callerid]
exten => 800,1,set(callerid=${CALLERID(num)})
same = n,agi(googletts.agi,"Votre numéro est le "${callerid} ". Bonne journée.",fr,any,1.5)
same = n,Hangup()
[soundtest]
exten => 802,1,Answer()
same = n,agi(googletts.agi,"Test de connexion réussi. Testez votre voix en parlant maintenant.",fr,any,1.5)
same = n,echo()
same = n,Hangup()
[get-time]
exten => 801,1,set(currentTime=${STRFTIME(${EPOCH},GMT-2,%H:%M)})
same = n,agi(googletts.agi,"Il est actuellement " ${currentTime},fr,any,1.5)
same = n,Hangup()
[ippi]
exten => s,1,PlayBack(custom/welcome)
same = n,Ringing
same = n,Dial(PJSIP/1067&PJSIP/1092,60,tTm(predecrochage))
same = n,PlayBack(custom/noagent)
same = n,HangUp()
[DialEndpoints] ; Internal calls - The extension I want to install Instant Messaging
exten => _10.,1,Ringing
same = n,Dial(PJSIP/${EXTEN},30,tTm(predecrochage))
same = n,PlayBack(custom/agentunavailable)
same = n,HangUp()
[live]
exten => 805,1,ConfBridge(confLive,live,user_live)
[unallowed]
exten => _.,1,PlayBack(/var/lib/asterisk/sounds/custom/forbidden)
[freecalls]
exten => _+1800.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1888.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1877.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1866.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1855.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1844.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _+1833.,1,Dial(PJSIP/${EXTEN}@voipms)
[Posteaposte]
include => DialEndpoints
include => get-time
include => soundtest
include => freecalls
include => unallowed
[Tout]
include => DialEndpoints
include => live
include => get-callerid
include => get-time
include => soundtest
exten => _+.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
include => unallowed
[Francais]
include => DialEndpoints
include => get-callerid
include => get-time
include => soundtest
exten => _+33.,1,Dial(PJSIP/${EXTEN}@voipms)
include => freecalls
include => unallowed
[FixesFrancais]
include => DialEndpoints
include => get-callerid
include => get-time
include => soundtest
include => freecalls
exten => _+33[1-59].,1,Dial(PJSIP/${EXTEN}@voipms)
include => unallowed
What should I change / install in Asterisk ?
Thanks for your help!