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