added some modules

root/app/start.sh

@@ -12,7 +12,7 @@ protocol=udp
 bind=0.0.0.0
 ; localnet=172.17.0.0/16
 external_media_address=${EXTERNAL_IP}
-external_signaling_address=${EXTERNAL_IP} 
+external_signaling_address=${EXTERNAL_IP}
 
 [mytrunk]
 type=registration
@@ -20,17 +20,17 @@ transport=mytransport
 outbound_auth=mytrunk
 server_uri=sip:${SIP_SERVER}:5060
 client_uri=sip:${SIP_USER}@${SIP_SERVER}:5060
- 
+
 [mytrunk]
 type=auth
 auth_type=userpass
 password=${SIP_PASS}
 username=${SIP_USER}
- 
+
 [mytrunk]
 type=aor
 contact=sip:${SIP_SERVER}:5060
- 
+
 [mytrunk]
 type=endpoint
 context=incoming-calls
@@ -38,11 +38,11 @@ disallow=all
 allow=speex,g726,g722,ilbc,gsm,alaw
 outbound_auth=mytrunk
 aors=mytrunk
- 
+
 [mytrunk]
 type=identify
 endpoint=mytrunk
 match=${SIP_SERVER}
 EOF
 
-/usr/sbin/asterisk -cv
+/usr/sbin/asterisk -cv

root/etc/asterisk/modules.conf

@@ -2,9 +2,12 @@
 autoload = no
 
 load = app_record.so
+load = app_mixmonitor.so
 load = app_verbose.so
 load = cdr_custom.so
 load = chan_pjsip.so
+load = app_waitforsilence.so
+load = app_playback.so
 
 ; Codecs
 
@@ -39,6 +42,8 @@ load = func_sorcery.so
 load = func_devstate.so
 load = func_strings.so
 load = func_env.so 
+load = func_shell.so
+
 
 load = pbx_config.so