KI-AGENT: Asterisk-Stand ohne FreePBX versionieren

telephony/asterisk/pjsip.conf

@@ -31,6 +31,7 @@ force_rport=yes
 rewrite_contact=yes
 rtp_symmetric=yes
 transport=transport-ws
+#tryinclude generated/pjsip.webrtc.conf
 from_domain=localhost
 
 [1001](fedeo-webrtc)

telephony/asterisk/rtp.conf

@@ -1,5 +1,5 @@
 [general]
 rtpstart=10000
-rtpend=10020
+rtpend=10100
 icesupport=yes
 strictrtp=no

telephony/freepbx/init.sql

@@ -1,7 +0,0 @@
-CREATE DATABASE IF NOT EXISTS asterisk;
-GRANT ALL PRIVILEGES ON `asterisk`.* TO 'freepbxuser'@'%';
-
-CREATE DATABASE IF NOT EXISTS asteriskcdrdb;
-GRANT ALL PRIVILEGES ON `asteriskcdrdb`.* TO 'freepbxuser'@'%';
-
-FLUSH PRIVILEGES;

telephony/freepbx/my.cnf

@@ -1,2 +0,0 @@
-[mysqld]
-sql_mode=NO_ENGINE_SUBSTITUTION

telephony/freepbx/secrets/freepbxuser_password.txt

@@ -1 +0,0 @@
-change-this-freepbx-user-password

telephony/freepbx/secrets/mysql_root_password.txt

@@ -1 +0,0 @@
-change-this-freepbx-root-password

telephony/freepbx/secrets/sasl_passwd.txt

@@ -1 +0,0 @@
-[smtp.example.com]:587 user@example.com:change-this-app-password

telephony/render-asterisk-config.sh

@@ -6,6 +6,7 @@ mkdir -p "$GENERATED_DIR"
 
 PJSIP_FILE="$GENERATED_DIR/pjsip.telekom.conf"
 EXTENSIONS_FILE="$GENERATED_DIR/extensions.telekom.conf"
+PJSIP_WEBRTC_FILE="$GENERATED_DIR/pjsip.webrtc.conf"
 
 enabled="${TELEPHONY_TELEKOM_ENABLED:-false}"
 
@@ -21,6 +22,9 @@ if [ "$enabled" != "true" ] && [ "$enabled" != "1" ] && [ "$enabled" != "yes" ];
 EOF
   cat > "$EXTENSIONS_FILE" <<'EOF'
 ; Telekom-Anbindung ist deaktiviert.
+EOF
+  cat > "$PJSIP_WEBRTC_FILE" <<'EOF'
+; Kein externes WebRTC-Media-Rewrite konfiguriert.
 EOF
   exit 0
 fi
@@ -32,6 +36,7 @@ password="${TELEPHONY_TELEKOM_PASSWORD:-}"
 caller_id="${TELEPHONY_TELEKOM_CALLER_ID:-$sip_user}"
 inbound_extension="${TELEPHONY_TELEKOM_INBOUND_EXTENSION:-1001}"
 outbound_prefix="${TELEPHONY_TELEKOM_OUTBOUND_PREFIX:-0}"
+external_media_address="${TELEPHONY_ASTERISK_EXTERNAL_MEDIA_ADDRESS:-${TELEPHONY_ASTERISK_EXTERNAL_SIGNALING_ADDRESS:-}}"
 
 if [ -z "$sip_user" ] || [ -z "$password" ]; then
   cat > "$PJSIP_FILE" <<'EOF'
@@ -39,6 +44,9 @@ if [ -z "$sip_user" ] || [ -z "$password" ]; then
 EOF
   cat > "$EXTENSIONS_FILE" <<'EOF'
 ; Telekom-Anbindung ist aktiviert, aber nicht vollständig konfiguriert.
+EOF
+  cat > "$PJSIP_WEBRTC_FILE" <<'EOF'
+; Kein externes WebRTC-Media-Rewrite konfiguriert.
 EOF
   echo "FEDEO Telefonie: Telekom-Anbindung unvollständig, Trunk wird nicht erzeugt." >&2
   exit 0
@@ -108,6 +116,10 @@ exten => _+X.,1,NoOp(FEDEO ausgehend über Telekom: \${EXTEN})
  same => n,Hangup()
 
 [from-telekom]
+exten => $caller_id,1,NoOp(FEDEO eingehend über Telekom DID: \${EXTEN})
+ same => n,Dial(PJSIP/$inbound_extension,30)
+ same => n,Hangup()
+
 exten => s,1,NoOp(FEDEO eingehend über Telekom)
  same => n,Dial(PJSIP/$inbound_extension,30)
  same => n,Hangup()
@@ -117,4 +129,16 @@ exten => _X!,1,NoOp(FEDEO eingehend über Telekom: \${EXTEN})
  same => n,Hangup()
 EOF
 
+if [ -n "$external_media_address" ]; then
+  cat > "$PJSIP_WEBRTC_FILE" <<EOF
+; Automatisch aus Umgebungsvariablen erzeugt. Nicht in Git einchecken.
+media_address=$external_media_address
+bind_rtp_to_media_address=yes
+EOF
+else
+  cat > "$PJSIP_WEBRTC_FILE" <<'EOF'
+; Kein externes WebRTC-Media-Rewrite konfiguriert.
+EOF
+fi
+
 echo "FEDEO Telefonie: Telekom-Trunk für $sip_user@$registrar erzeugt."

telephony/vps-asterisk.env

@@ -5,6 +5,8 @@ TELEPHONY_ENABLED=true
 TELEPHONY_ASTERISK_HTTP_URL=http://188.245.76.1:8088/ws
 TELEPHONY_ASTERISK_WS_URL=ws://188.245.76.1:8088/ws
 TELEPHONY_SIP_DOMAIN=188.245.76.1
+TELEPHONY_ASTERISK_EXTERNAL_SIGNALING_ADDRESS=188.245.76.1
+TELEPHONY_ASTERISK_EXTERNAL_MEDIA_ADDRESS=188.245.76.1
 TELEPHONY_ASTERISK_AMI_HOST=127.0.0.1
 TELEPHONY_ASTERISK_AMI_PORT=5038
 TELEPHONY_ASTERISK_AMI_USER=fedeo