File sip_peer_status-18.patch of Package asterisk

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File sip_peer_status-18.patch of Package asterisk

--- asterisk-18.17.1/channels/chan_sip.c	2023-04-03 11:47:51.000000000 -0400
+++ asterisk-18.17.1-new/channels/chan_sip.c	2023-04-04 16:08:16.661189344 -0400
@@ -4156,6 +4156,21 @@
 				pkt->is_resp ? "Response" : "Request",
 				(int) ast_tvdiff_ms(ast_tvnow(), pkt->time_sent));
 		}
+		if (pkt->is_fatal) {
+			const struct ast_sockaddr *real_dst = sip_real_dst(pkt->owner);
+			manager_event(EVENT_FLAG_SYSTEM, "SIPCriticalTimeout",
+				"Type: %s\r\n"
+				"CallID: %s\r\n"
+				"SeqNo: %u\r\n"
+				"Host: %s\r\n"
+				"Timeout: %dms\r\n",
+				pkt->is_resp ? "Response" : "Request",
+				pkt->owner->callid,
+				pkt->seqno,
+				ast_sockaddr_stringify(real_dst),
+				(int) ast_tvdiff_ms(ast_tvnow(), pkt->time_sent)
+			);
+		}
 	} else if (pkt->method == SIP_OPTIONS && sipdebug) {
 		ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s)  -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n", pkt->owner->callid);
 	}
@@ -6631,6 +6646,22 @@
 								dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
 								dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
 								dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
+
+		if ( p->owner ) {
+			manager_event(EVENT_FLAG_CALL, "SIPInvite",
+				"Channel: %s\r\n"
+				"Uniqueid: %s\r\n"
+				"SIPCallID: %s\r\n"
+				"CallerIDName: %s\r\n"
+				"ConnectedLineName: %s\r\n",
+				ast_channel_name(p->owner),
+				ast_channel_uniqueid(p->owner),
+				p->callid,
+				ast_channel_caller(p->owner)->id.name.str,
+				ast_channel_connected(p->owner)->id.name.str
+			);
+		}
+
 		sip_pvt_unlock(p);
 	}
 	return res;
@@ -17185,6 +17216,13 @@
 	if (ast_sockaddr_cmp(&peer->addr, &oldsin)) {
 		ast_verb(3, "Registered SIP '%s' at %s\n", peer->name,
 			ast_sockaddr_stringify(&peer->addr));
+		manager_event(EVENT_FLAG_SYSTEM, "PeerRegistered",
+			"ChannelType: SIP\r\n"
+                        "Peer: SIP/%s\r\n"
+                        "Host: %s\r\n",
+                        peer->name,
+			ast_sockaddr_stringify(&peer->addr)
+                );
 	}
 	sip_pvt_unlock(pvt);
 	sip_poke_peer(peer, 0);
@@ -25008,6 +25046,22 @@
 
 		ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
 			peer->name, s, pingtime, peer->maxms);
+
+		manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
+			"ChannelType: SIP\r\n"
+                        "Peer: SIP/%s\r\n"
+                        "PeerStatus: %s\r\n"
+			"Time: %d\r\n"
+                        "PingTime: %dms\r\n"
+                        "MaxPing: %dms\r\n",
+                        peer->name,
+			s,
+			peer->lastms,
+			pingtime,
+                        peer->maxms
+                );
+
+		
 		ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
 		if (sip_cfg.peer_rtupdate) {
 			ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
@@ -25202,6 +25256,24 @@
 		}
 	}
 
+	if ( p->owner ) {
+		manager_event(EVENT_FLAG_CALL, "SIPResponse",
+			"Response: %d\r\n"
+			"Channel: %s\r\n"
+			"Uniqueid: %s\r\n"
+			"SIPCallID: %s\r\n"
+			"CallerIDName: %s\r\n"
+			"ConnectedLineName: %s\r\n",
+ 			resp,
+			ast_channel_name(p->owner),
+			ast_channel_uniqueid(p->owner),
+			p->callid,
+			ast_channel_caller(p->owner)->id.name.str,
+			ast_channel_connected(p->owner)->id.name.str
+			);
+	}
+
+
 	/* Acknowledge whatever it is destined for */
 	if ((resp >= 100) && (resp <= 199)) {
 		/* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
@@ -29986,6 +30058,16 @@
 				}
 				ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
 					ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx));
+
+				manager_event(EVENT_FLAG_CALL, "SIPRTPDisconnect",
+					"Channel: %s\r\n"
+					"RTPLastRX: %ld\r\n",
+					ast_channel_name(dialog->owner),
+					(long) dialog->lastrtprx
+				);
+
+
+
 				send_session_timeout(dialog->owner, "RTPTimeout");
 
 				/* Issue a softhangup - cause 44 (as used by Cisco for RTP timeouts) */
@@ -30549,6 +30631,19 @@
 	if (peer->lastms > -1) {
 
 		ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE!  Last qualify: %d\n", peer->name, peer->lastms);
+
+		manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
+			"ChannelType: SIP\r\n"
+			"Peer: SIP/%s\r\n"
+			"PeerStatus: Unreachable\r\n"
+			"Time: %d\r\n"
+			"PingTime: -1ms\r\n"
+			"MaxPing: %dms\r\n",
+			peer->name,
+			peer->lastms,
+			peer->maxms
+		);
+
 		if (sip_cfg.peer_rtupdate) {
 			ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", "-1", SENTINEL);
 		}

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File sip_peer_status-18.patch of Package asterisk

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