#!/bin/bash
# (C)2009 DatuX - Released under GPL

# Asterisk test script
# Test de instellingen die gemaakt zijn door 01100_asteriskconf uit via de pjsip SIP client en via ssh commandos

#dependencys:
# -PBX op 192.168.13.1 met sip account 650, password 'test'. nummer 111 moet automatisch antwoorden op deze pbx.
# -geluidskaart waarvan output terug gekoppeld staat naar capture input
# -geluidskaart moet multichannel zijn (dus meerdere dingen tegelijk afspelen)
# -om filmpjes te maken: xvidcap starten met 'aoss xvidcap'

source testlib || exit 1

DESCRIPTION1="Sla test over indien we niet de juiste licentieoptie hebben."
CMD="cat /etc/webint/REG"
VERIFYPOSITIVE='"VAGU"'
if ! sshtest; then
	echo "Dit is geen voiceserver, test overgeslagen"
	exit 0
fi



TIMEOUT=30

##################################################### Test afdeling #############################
# Tests hieronder toevoegen, gebruik makend van de functies van asterisktestlib



info "Basic check: Geen warning en errors in de logs bij herstart met lege logs?";
CMD="svcstop /service/asterisk
rm /var/log/asterisk/full
svcstart /service/asterisk
"
sshtest || exit 1
asterisk_errorcheck

info "Basic check: Klopt het aantal geladen modules?";
VERIFYPOSITIVE="157 modules loaded"
asterisk_cmd "show modules" || exit 1


#verbose mode
VERIFYPOSITIVE="Verbosity"
asterisk_cmd "set verbose 100" || exit 1
VERIFYPOSITIVE="Core debug"
asterisk_cmd "set debug 100" || exit 1


info "Pjsua starten"
pjsua_restart


####################################### SIP: registratie sip accounts
info "Register user 100 tot 105"
for USER in `seq 100 105`; do
	NOWAIT=1
	PASSWD="test"
	sip_register $USER || exit 1
done


####################################### SIP: basic checks om te zien of het uberhaubt werkt
info "Basic check: Bel 100 en hang op"
sip_call 100 || exit 1
sip_hangup || exit 1


info "Basic check: bel 100, en beantwoord en hangop"
sip_call 101 || exit 1
sip_answer || exit 1
sip_hangup || exit 1



####################################### SIP: voicemail systeem checken
DESCRIPTION1="voorbereiding voicemail test: verwijder alle voicemail zooi 105 van server"
CMD="rm -r /var/spool/asterisk/voicemail/default/105"
sshtest || exit 1

info "Stel unavailable message in"
sip_call '#' || exit 1
sip_waitanswer || exit 1
sip_dtmf "0" || exit 1
sip_dtmf "1" || exit 1
info "Spreek unavailable message in..."
sleep 4
makesound
sip_dtmf "#" || exit 1
sip_dtmf "1" || exit 1
sip_hangup || exit 1

DESCRIPTION1="unavail nu aanwezig op server?"
CMD="ls /var/spool/asterisk/voicemail/default/105"
VERIFYPOSITIVE="unavail.wav"
sshtest || exit 1

info "Bel 105 en laat bericht achter"
sip_call 105 || exit 1
sip_waitanswer || exit 1
sleep 6
info "Spreek bericht in..."
makesound
makesound
sip_dtmf '#'
sip_hangup || exit 1

DESCRIPTION1="check op server of we echt voicemail hebben. er moet wel geluid zijn ontvangen, anders drop asterisk het mailtje! "
CMD="ls /var/spool/asterisk/voicemail/default/105/INBOX"
VERIFYPOSITIVE="wav"
sshtest || exit 1


info "Bel voicemail 105, luister het mailtje en maak hem leeg"
sip_call '#' || exit 1
sip_waitanswer || exit 1
sip_dtmf "1" || exit 1
sleep 3
sip_dtmf "7" || exit 1
sip_hangup || exit 1

DESCRIPTION1="is voicemail box echt leeg?"
CMD="ls /var/spool/asterisk/voicemail/default/105/INBOX"
VERIFYNEGATIVE="wav"
sshtest || exit 1


####################################### SIP: follow me
info "Followme testen"
#extentie 111 draait op de datux centrale en moet meteen opnemen. (staat nu op 'directory service' insgesteld')
sip_call 103 || exit 1
#wacht todat de incomende call op 103 stop. vanaf dat moment word followme actief:
sip_waithangup || exit 1
sleep 2
VERIFYPOSITIVE=" FollowMe"
asterisk_cmd "show channels" || exit 1
sip_hangup || exit 1



####################################### SIP: voip trunk
info "Voip trunk testen"
#extentie 111 draait op de datux centrale en moet meteen opnemen. (staat nu op 'directory service' insgesteld')
sip_call 111 || exit 1
sip_waitanswer || exit 1
sip_hangup || exit 1




####################################### SIP: voice menu
info "Voicemenu testen (menu 600 geeft piep en schakeld door naar 100)"
sip_call 600 || exit 1
sip_waitanswer || exit 1
sip_answer || exit 1
sip_hangup || exit 1



####################################### SIP: call queue
DESCRIPTION1="niet ingelogd op queue?"
VERIFYPOSITIVE="Agent/105.*(Invalid|has taken no calls yet)"
asterisk_cmd "show queue" || exit 1

info "Login op via extentie 1000"
sip_call 1000 || exit 1
sip_waitanswer || exit 1
sip_dtmf '105#' || exit 1
sleep 2

DESCRIPTION1="wel ingelogd op queue?"
VERIFYPOSITIVE="Agent/105.*Not in use"
asterisk_cmd "show queue" || exit 1

info "Bel naar queue"
sip_call 800 || exit 1
sip_waitanswer || exit 1

DESCRIPTION1="agent is nu busy?"
VERIFYPOSITIVE="Agent/105.*Busy"
asterisk_cmd "show queue" || exit 1

makesound

sip_hangup || exit 1

DESCRIPTION1="niet ingelogd op queue?"
VERIFYPOSITIVE="Agent/105.*Unavailable.*last was"
asterisk_cmd "show queue" || exit 1



###################################### SIP: meet me

#asterisk 1.4.23.2 segfault hier op!
#1.4.22.2 doet het wel.
sip_call 500 || exit 1
sip_waitanswer || exit 1
sleep 2
makesound
sip_dtmf '#' || exit 1

sip_call 500 || exit 1
sip_waitanswer || exit 1
sleep 2
makesound
sip_dtmf '#' || exit 1

info "Conference is now active.."
makesound

VERIFYPOSITIVE="2 users in that conference"
asterisk_cmd "meetme list 500" 

sip_hangup || exit 1


####################################### SIP: call transfer en music on hold testen
info "bel 100, en transfer naar 101"
sip_call 100 || exit 1
sip_answer || exit 1
sip_transfer 101 || exit 1
sip_answer || exit 1
sip_hangup || exit 1

info "bel 100, en beantwoord en zet in hold"
sip_call 100 || exit 1
sip_answer || exit 1
sip_hold || exit 1

sleep 2

info "Lijkt de music on hold het te doen?"
CMD="lsof /var/lib/asterisk/moh/*"
VERIFYPOSITIVE="wav"
sshtest || exit 1

info "Unhold en hang op"
sip_unhold || exit 1
sip_hangup || exit 1


####################################### SIP: ring groups
info "Test ringgroups "
sip_call 700 || exit 1
VERIFYPOSITIVE="4 active channels"
asterisk_cmd "show channels" || exit 1
sip_hangup || exit 1
VERIFYPOSITIVE="0 active channels"
asterisk_cmd "show channels" || exit 1


#laatste fout-controle na alle tests:
asterisk_errorcheck


info "Regression test completed successfully"