--- asterisk/channels/chan_sip.c	2005-01-27 22:38:16.000000000 -0700
+++ asterisk-20050129/channels/chan_sip.c	2005-01-29 16:25:18.000000000 -0700
@@ -401,6 +401,7 @@
 	struct sip_history *history;		/* History of this SIP dialog */
 	struct ast_variable *vars;
 	struct sip_pvt *next;			/* Next call in chain */
+	int onhold;				/* call on hold */
 } *iflist = NULL;
 
 #define FLAG_RESPONSE (1 << 0)
@@ -586,6 +587,7 @@
 static void prune_peers(void);
 static int sip_do_reload(void);
 
+static int holdevents = 0;
 
 /*--- sip_debug_test_addr: See if we pass debug IP filter */
 static inline int sip_debug_test_addr(struct sockaddr_in *addr) 
@@ -2803,10 +2805,25 @@
 			/* Turn on/off music on hold if we are holding/unholding */
 			if (sin.sin_addr.s_addr && !sendonly) {
 				ast_moh_stop(ast_bridged_channel(p->owner));
+				if (holdevents && p->onhold) {
+					manager_event(EVENT_FLAG_CALL, "Unhold",
+							"Channel: %s\r\n"
+							"Uniqueid: %s\r\n",
+							p->owner->name, 
+							p->owner->uniqueid);
+				}
 			} else {
 				ast_moh_start(ast_bridged_channel(p->owner), NULL);
 				if (sendonly)
 					ast_rtp_stop(p->rtp);
+				if (holdevents) {
+					manager_event(EVENT_FLAG_CALL, "Hold",
+							"Channel: %s\r\n"
+							"Uniqueid: %s\r\n",
+							p->owner->name, 
+							p->owner->uniqueid);
+					p->onhold = 1;
+				}
 			}
 		}
 	}
@@ -9229,6 +9246,8 @@
 			} else {
 				ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config);
 			}
+		} else if (!strcasecmp(v->name, "holdevents")) {
+			holdevents = ast_true(v->value);
 		}
 		/* else if (strcasecmp(v->name,"type"))
 		 *	ast_log(LOG_WARNING, "Ignoring %s\n", v->name);
--- asterisk/configs/sip.conf.sample	2005-01-11 11:39:48.000000000 -0700
+++ asterisk-20050129/configs/sip.conf.sample	2005-01-29 16:21:39.000000000 -0700
@@ -118,6 +118,7 @@
 ;               (instead of type=friend) if you have calls in both directions
   
 ;registertimeout=20		; retry registration calls every 20 seconds (default)
+;holdevents=no			; generate manager events when sip ua places call on hold
 
 ;---------------------------------------------- NAT SUPPORT ------------------------
 ; The externip, externhost and localnet settings are used if you use Asterisk behind