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Merge callevents etc (bug #3456)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4921 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-01-29 23:19:01 +00:00
parent 48dd0fef1d
commit 8f199888d5
2 changed files with 23 additions and 1 deletions
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23
channels/chan_sip.c
View File

@@ -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);

1
configs/sip.conf.sample
View File

@@ -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