Merge callevents etc (bug #3456)
authorMark Spencer <markster@digium.com>
Sat, 29 Jan 2005 23:19:01 +0000 (23:19 +0000)
committerMark Spencer <markster@digium.com>
Sat, 29 Jan 2005 23:19:01 +0000 (23:19 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4921 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c
configs/sip.conf.sample

index 3132330..ceeb3e5 100755 (executable)
@@ -388,7 +388,7 @@ static struct sip_pvt {
        int rtpkeepalive;                       /* Send RTP packets for keepalive */
 
        int subscribed;                         /* Is this call a subscription?  */
-       int stateid;
+       int stateid;
        int dialogver;
        
        struct ast_dsp *vad;
@@ -401,6 +401,7 @@ static struct sip_pvt {
        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 int update_user_counter(struct sip_pvt *fup, int event);
 static void prune_peers(void);
 static int sip_do_reload(void);
 
+static int callevents = 0;
 
 /*--- sip_debug_test_addr: See if we pass debug IP filter */
 static inline int sip_debug_test_addr(struct sockaddr_in *addr) 
@@ -2803,7 +2805,23 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                        /* 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 (callevents && p->onhold) {
+                                       manager_event(EVENT_FLAG_CALL, "Unhold",
+                                               "Channel: %s\r\n"
+                                               "Uniqueid: %s\r\n",
+                                               p->owner->name, 
+                                               p->owner->uniqueid);
+                                       p->onhold = 0;
+                               }
                        } else {
+                               if (callevents && !p->onhold) {
+                                       manager_event(EVENT_FLAG_CALL, "Hold",
+                                               "Channel: %s\r\n"
+                                               "Uniqueid: %s\r\n",
+                                               p->owner->name, 
+                                               p->owner->uniqueid);
+                                               p->onhold = 1;
+                               }
                                ast_moh_start(ast_bridged_channel(p->owner), NULL);
                                if (sendonly)
                                        ast_rtp_stop(p->rtp);
@@ -9065,6 +9083,7 @@ static int reload_config(void)
        videosupport = 0;
        compactheaders = 0;
        relaxdtmf = 0;
+       callevents = 0;
        ourport = DEFAULT_SIP_PORT;
        global_rtptimeout = 0;
        global_rtpholdtimeout = 0;
@@ -9229,6 +9248,8 @@ static int reload_config(void)
                        } 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, "callevents")) {
+                       callevents = ast_true(v->value);
                }
                /* else if (strcasecmp(v->name,"type"))
                 *      ast_log(LOG_WARNING, "Ignoring %s\n", v->name);
index 7ba7b53..0506173 100755 (executable)
@@ -118,6 +118,7 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
 ;               (instead of type=friend) if you have calls in both directions
   
 ;registertimeout=20            ; retry registration calls every 20 seconds (default)
+;callevents=no                 ; generate manager events when sip ua performs events (e.g. hold)
 
 ;---------------------------------------------- NAT SUPPORT ------------------------
 ; The externip, externhost and localnet settings are used if you use Asterisk behind