mirror of
https://github.com/iiab/iiab.git
synced 2025-03-09 15:40:17 +00:00
asterisk-test-script
A simple and basic script that creates extensions and a quick call from the console to help the user quickly test if asterisk is working (could add more tests in future).
This commit is contained in:
lemueldsouza
parent
977c706024
commit
baca7e9386
3 changed files with 308 additions and 0 deletions
245
roles/pbx/templates/asterisk_test
Normal file
245
roles/pbx/templates/asterisk_test
Normal file
|
|
@ -0,0 +1,245 @@
|
|||
#!/bin/bash
|
||||
|
||||
#
|
||||
# This script creates two extensions (301 and 302)
|
||||
# Credentials can be found in pjsip_custom.conf
|
||||
# For regular implementation, you are requested to create extensions using the FreePBX webpage to avoid any issues
|
||||
#
|
||||
# Usage:
|
||||
# sudo ./asterisk_test
|
||||
# Default option
|
||||
# Reverts file changes done during the test and restarts asterisk towards the end
|
||||
#
|
||||
# sudo./asterisk_test retain
|
||||
# Files not reverted after the test. Extensions created remain active
|
||||
#
|
||||
# sudo./asterisk_test revert
|
||||
# This option exists in case you selected retain earlier, but now like to revert the changes
|
||||
# This only reverts the changes to the files and restarts asterisk, no other actions are performed
|
||||
#
|
||||
# sudo./asterisk_test restart
|
||||
# Restarts asterisk, no other actions are performed
|
||||
#
|
||||
# sudo./asterisk_test testcall
|
||||
# Makes a test call from asterisk console to extension 1000 which just responds with some audio
|
||||
#
|
||||
# Algo:
|
||||
# Please run the script as root
|
||||
#
|
||||
# 1. First check that asterisk -rx "pjsip show endpoints" returns no extensions as we haven't created any yet
|
||||
#
|
||||
# 2. Take a backup of existing files
|
||||
# * Rename existing pjsip_custom.conf at /etc/asterisk/ to pjsip_custom.freepbx.conf
|
||||
# * Rename existing extensions_custom.conf at /etc/asterisk/ to extensions_custom.freepbx.conf
|
||||
#
|
||||
# 3. Copy files pjsip_custom.conf and extensions_custom.conf provided with this script to /etc/asterisk
|
||||
#
|
||||
# 4. Change file permissions to asterisk:asterisk
|
||||
#
|
||||
# 5. Run fwconsole restart so that it picks up the new confs
|
||||
#
|
||||
# 6. asterisk -rx "pjsip show endpoints" should now show the extensions created
|
||||
#
|
||||
# 7. Make a call from asterisk console to the internal extension 1000. This extension is only used to respond
|
||||
# with an audio message, no need to register this extension.
|
||||
# * Check asterisk logs at /var/log/asterisk/full to see if you can see information about calls to
|
||||
# extension 1000 or to context iiab-test or check if the any of the playback files are executed
|
||||
# * If found, test is successful
|
||||
#
|
||||
# 8. Once done with the experiment, delete the two newly created files and
|
||||
# * rename pjsip_custom.freepbx.conf to pjsip_custom.conf and
|
||||
# * rename extensions_custom.freepbx.conf to extensions_custom.conf
|
||||
#
|
||||
# 9. A better test would be to register the extension using your softphone app (using Linphone android app for this example)
|
||||
# * Register the created extension on a softphone with the help of PBX README (check the credentials in pjsip_custom.conf)
|
||||
# * Dial '1000' and hear the automated response
|
||||
# * Or Dial the other extension that you created if you have registered two extensions
|
||||
#
|
||||
|
||||
|
||||
ROOT_UID=0
|
||||
E_NOTROOT=67
|
||||
AST_DIR=/etc/asterisk
|
||||
AST_LOG_FILE=/var/log/asterisk/full
|
||||
PJSIP_CUST_CONF=pjsip_custom.conf
|
||||
PJSIP_CUST_CONF_BKUP=pjsip_custom.freepbx.conf
|
||||
EXT_CUST_CONF=extensions_custom.conf
|
||||
EXT_CUST_CONF_BKUP=extensions_custom.freepbx.conf
|
||||
SCRIPT_ARG=$1
|
||||
|
||||
#
|
||||
# Check if extensions that you created exist
|
||||
#
|
||||
function check_if_extensions_exist() {
|
||||
echo -e "\n${FUNCNAME[0]}(): Checking if test extension exists..."
|
||||
extn_exists=`asterisk -rx 'pjsip show endpoints'|grep '301'`
|
||||
if [ -z "$extn_exists" ]
|
||||
then
|
||||
echo -e "${FUNCNAME[0]}(): Test extension does not exist"
|
||||
else
|
||||
echo -e "${FUNCNAME[0]}(): Test extension exists"
|
||||
fi
|
||||
}
|
||||
|
||||
#
|
||||
# Copy files to AST_DIR for testing
|
||||
#
|
||||
function copy_files_for_test() {
|
||||
echo -e "\n${FUNCNAME[0]}(): Copying files for testing..."
|
||||
|
||||
# Proceed if source files exist in pwd
|
||||
if [[ -f "${PJSIP_CUST_CONF}" && -f "${EXT_CUST_CONF}" ]]
|
||||
then
|
||||
# Rename original files
|
||||
mv ${AST_DIR}/${PJSIP_CUST_CONF} ${AST_DIR}/${PJSIP_CUST_CONF_BKUP}
|
||||
mv ${AST_DIR}/${EXT_CUST_CONF} ${AST_DIR}/${EXT_CUST_CONF_BKUP}
|
||||
|
||||
# Copy files supplied with the script to destination and change their owner and permissions
|
||||
cp ${PJSIP_CUST_CONF} ${EXT_CUST_CONF} ${AST_DIR}
|
||||
chown asterisk:asterisk ${AST_DIR}/${PJSIP_CUST_CONF} ${AST_DIR}/${EXT_CUST_CONF}
|
||||
chmod ug=rw,o=r ${AST_DIR}/${PJSIP_CUST_CONF} ${AST_DIR}/${EXT_CUST_CONF}
|
||||
else
|
||||
echo -e "\n${FUNCNAME[0]}(): Files ${PJSIP_CUST_CONF} and ${EXT_CUST_CONF} do not exist in pwd. Exiting!!!"
|
||||
exit 1
|
||||
fi
|
||||
}
|
||||
|
||||
#
|
||||
# Check if test call was successful
|
||||
#
|
||||
function check_call_success() {
|
||||
echo -e "\n${FUNCNAME[0]}(): Making a test call to extension 1000..."
|
||||
|
||||
# Make a call from asterisk to extension 1000 to receive automated response
|
||||
asterisk -rx 'console dial 1000@iiab-test'
|
||||
|
||||
# This may not be the best way in case you plan to run the script multiple times
|
||||
# or if the script is run at the end of the hour, but since this will be run
|
||||
# as a basic test after first time freepbx install, this should work.
|
||||
# Feel free to try a better search or add minute check - $(date +"%Y-%m-%d %H:%M")
|
||||
# if you find that better
|
||||
test_run=`grep "$(date +'%Y-%m-%d %H')" ${AST_LOG_FILE}|grep "Playing 'goodbye"`
|
||||
|
||||
if [ -z "$test_run" ]
|
||||
then
|
||||
echo -e "${FUNCNAME[0]}(): Test call to extension 1000 not successful"
|
||||
else
|
||||
echo -e "${FUNCNAME[0]}(): Test call to extension 1000 successful"
|
||||
fi
|
||||
}
|
||||
|
||||
#
|
||||
# Restart asterisk and make sure it's running
|
||||
#
|
||||
function restart_asterisk() {
|
||||
echo -e "\n${FUNCNAME[0]}(): Restarting asterisk..."
|
||||
|
||||
# There should be a better way than a fwconsole restart, but for now this works
|
||||
# If you haven't installed FreePBX, use systemctl restart asterisk
|
||||
# It doesn't work so well, so you may have to execute it twice
|
||||
fwconsole restart
|
||||
|
||||
sleep 5
|
||||
|
||||
# Occasionally displays 3 or 4 during tests, the old process takes time to exit
|
||||
no_of_astersisk_procs=`pgrep -c asterisk`
|
||||
echo -e "${FUNCNAME[0]}(): No of asterisk procs: ${no_of_astersisk_procs}"
|
||||
}
|
||||
|
||||
#
|
||||
# Revert file changes
|
||||
#
|
||||
function revert_file_changes() {
|
||||
|
||||
# Do this only if extensions_custom.freepbx.conf and pjsip_custom.freepbx.conf exist
|
||||
if [[ -f "${AST_DIR}/${PJSIP_CUST_CONF_BKUP}" && -f "${AST_DIR}/${EXT_CUST_CONF_BKUP}" ]]
|
||||
then
|
||||
echo -e "\n${FUNCNAME[0]}(): Reverting file changes"
|
||||
rm ${AST_DIR}/${PJSIP_CUST_CONF} ${AST_DIR}/${EXT_CUST_CONF}
|
||||
mv ${AST_DIR}/${EXT_CUST_CONF_BKUP} ${AST_DIR}/${EXT_CUST_CONF}
|
||||
mv ${AST_DIR}/${PJSIP_CUST_CONF_BKUP} ${AST_DIR}/${PJSIP_CUST_CONF}
|
||||
else
|
||||
echo -e "\n${FUNCNAME[0]}(): Nothing to revert - Files ${PJSIP_CUST_CONF_BKUP} and ${EXT_CUST_CONF_BKUP} do not exist in ${AST_DIR} Exiting!!!"
|
||||
exit 1
|
||||
fi
|
||||
}
|
||||
|
||||
#
|
||||
# Revert file changes and restart asterisk
|
||||
#
|
||||
function revert_changes_and_restart_asterisk() {
|
||||
if [ "$SCRIPT_ARG" == "retain" ]
|
||||
then
|
||||
echo -e "\n${FUNCNAME[0]}(): User decided to retain changes done during the test..."
|
||||
else
|
||||
# Default - revert changes
|
||||
echo -e "\n${FUNCNAME[0]}(): Reverting file changes done during the test and restarting asterisk to get back to original state..."
|
||||
|
||||
revert_file_changes
|
||||
restart_asterisk
|
||||
fi
|
||||
}
|
||||
|
||||
#
|
||||
# Script usage
|
||||
#
|
||||
function script_usage() {
|
||||
echo -e "sudo ./asterisk_test"
|
||||
echo -e "\tDefault - Reverts file changes done during the test and restarts asterisk"
|
||||
echo -e "sudo ./asterisk_test retain"
|
||||
echo -e "\tFiles not reverted after the test. Extensions created remain active"
|
||||
echo -e "sudo ./asterisk_test revert"
|
||||
echo -e "\tIn case you selected retain earlier, but now like to revert the changes"
|
||||
echo -e "\tThis only reverts the changes to the files and restarts asterisk, no other actions are performed"
|
||||
echo -e "sudo ./asterisk_test restart"
|
||||
echo -e "\tRestarts asterisk, no other actions are performed"
|
||||
echo -e "sudo ./asterisk_test testcall"
|
||||
echo -e "\tMakes a test call to extension 1000"
|
||||
exit 1
|
||||
}
|
||||
|
||||
#
|
||||
# Main function that runs the script
|
||||
#
|
||||
function runscript() {
|
||||
echo -e "\n--------- Asterisk extension setup script - START --------------"
|
||||
|
||||
check_if_extensions_exist
|
||||
copy_files_for_test
|
||||
restart_asterisk
|
||||
check_if_extensions_exist
|
||||
|
||||
check_call_success
|
||||
revert_changes_and_restart_asterisk
|
||||
|
||||
echo -e "\n-------- Asterisk extension setup script - COMPLETE -------------"
|
||||
exit 0
|
||||
}
|
||||
|
||||
|
||||
|
||||
if [ "$UID" -ne "$ROOT_UID" ]
|
||||
then
|
||||
echo -e "\nSorry, you must be root to run this script."
|
||||
exit $E_NOTROOT
|
||||
fi
|
||||
|
||||
|
||||
case "$SCRIPT_ARG" in
|
||||
retain|"")
|
||||
runscript
|
||||
;;
|
||||
revert)
|
||||
revert_changes_and_restart_asterisk
|
||||
;;
|
||||
restart)
|
||||
restart_asterisk
|
||||
;;
|
||||
testcall)
|
||||
check_call_success
|
||||
;;
|
||||
*)
|
||||
script_usage
|
||||
exit 2
|
||||
;;
|
||||
esac
|
||||
19
roles/pbx/templates/extensions_custom.conf
Normal file
19
roles/pbx/templates/extensions_custom.conf
Normal file
|
|
@ -0,0 +1,19 @@
|
|||
;
|
||||
; Added for IIAB's FreePBX test
|
||||
; Refer asterisk documentation to for more details - https://wiki.asterisk.org/wiki/display/AST/Creating+Dialplan+Extensions
|
||||
;
|
||||
[general]
|
||||
[iiab-test]
|
||||
exten => 301,1,Dial(PJSIP/301)
|
||||
|
||||
exten => 302,1,Dial(PJSIP/302)
|
||||
|
||||
exten => 1000,1,Answer()
|
||||
same => n,Wait(1)
|
||||
same => n,Playback(sorry)
|
||||
same => n,Playback(you-have-reached-a-test-number)
|
||||
same => n,Playback(thanks-for-calling-today)
|
||||
same => n,Wait(1)
|
||||
same => n,Playback(goodbye)
|
||||
same => n,Hangup()
|
||||
|
||||
44
roles/pbx/templates/pjsip_custom.conf
Normal file
44
roles/pbx/templates/pjsip_custom.conf
Normal file
|
|
@ -0,0 +1,44 @@
|
|||
;
|
||||
; Added for IIAB's FreePBX test
|
||||
; Refer asterisk documentation to for more details - https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts
|
||||
;
|
||||
[transport-udp]
|
||||
type=transport
|
||||
protocol=udp
|
||||
bind=0.0.0.0
|
||||
|
||||
[301]
|
||||
type=endpoint
|
||||
context=iiab-test
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
auth=301-auth
|
||||
aors=301
|
||||
|
||||
[301-auth]
|
||||
type=auth
|
||||
auth_type=userpass
|
||||
username=301
|
||||
password=iiabtest
|
||||
|
||||
[301]
|
||||
type=aor
|
||||
max_contacts=1
|
||||
|
||||
[302]
|
||||
type=endpoint
|
||||
context=iiab-test
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
auth=302-auth
|
||||
aors=302
|
||||
|
||||
[302-auth]
|
||||
type=auth
|
||||
auth_type=userpass
|
||||
username=302
|
||||
password=iiabtest
|
||||
|
||||
[302]
|
||||
type=aor
|
||||
max_contacts=1
|
||||
Loading…
Add table
Reference in a new issue