Index: pbx.c
===================================================================
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.263
diff -u -r1.263 pbx.c
--- pbx.c 19 Jul 2005 15:54:17 -0000 1.263
+++ pbx.c 25 Jul 2005 18:52:34 -0000
@@ -1736,7 +1736,7 @@
char *cur, *rest;
int res = -1;
int allunavailable = 1, allbusy = 1, allfree = 1;
- int busy = 0;
+ int busy = 0, inuse = 0, ring = 0;
if (!e)
return -1;
@@ -1758,7 +1758,15 @@
allbusy = 0;
break;
case AST_DEVICE_INUSE:
- return AST_EXTENSION_INUSE;
+ inuse = 1;
+ allunavailable = 0;
+ allfree = 0;
+ break;
+ case AST_DEVICE_RINGING:
+ ring = 1;
+ allunavailable = 0;
+ allfree = 0;
+ break;
case AST_DEVICE_BUSY:
allunavailable = 0;
allfree = 0;
@@ -1777,7 +1785,13 @@
cur = rest;
} while (cur);
- if (allfree)
+ if (!inuse && ring)
+ return AST_EXTENSION_RINGING;
+ if (inuse && ring)
+ return AST_EXTENSION_RINGING_AND_INUSE;
+ if (inuse)
+ return AST_EXTENSION_INUSE;
+ if (allfree)
return AST_EXTENSION_NOT_INUSE;
if (allbusy)
return AST_EXTENSION_BUSY;
Index: channels/chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.787
diff -u -r1.787 chan_sip.c
--- channels/chan_sip.c 25 Jul 2005 17:39:48 -0000 1.787
+++ channels/chan_sip.c 25 Jul 2005 18:52:43 -0000
@@ -139,6 +139,11 @@
#define SIP_PUBLISH 14
#define SIP_RESPONSE 100
+#define SUBSCRIBE_XPIDF_XML 1
+#define SUBSCRIBE_DIALOG_INFO_XML 2
+#define SUBSCRIBE_CPIM_PIDF_XML 3
+#define SUBSCRIBE_PIDF_XML 4
+
#define RTP 1
#define NO_RTP 0
const struct cfsip_methods {
@@ -263,6 +268,7 @@
#define DEFAULT_CONTEXT "default"
static char default_context[AST_MAX_CONTEXT] = DEFAULT_CONTEXT;
+static char default_subscribecontext[AST_MAX_CONTEXT];
static char default_language[MAX_LANGUAGE] = "";
@@ -274,6 +280,7 @@
#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
static char default_notifymime[AST_MAX_EXTENSION] = DEFAULT_NOTIFYMIME;
+static int global_notifyringing = 1; /* Send notifications on ringing */
static int default_qualify = 0; /* Default Qualify= setting */
@@ -517,6 +524,7 @@
char from[256]; /* The From: header */
char useragent[256]; /* User agent in SIP request */
char context[AST_MAX_CONTEXT]; /* Context for this call */
+ char subscribecontext[AST_MAX_CONTEXT]; /* Subscribecontext */
char fromdomain[MAXHOSTNAMELEN]; /* Domain to show in the from field */
char fromuser[AST_MAX_EXTENSION]; /* User to show in the user field */
char fromname[AST_MAX_EXTENSION]; /* Name to show in the user field */
@@ -565,6 +573,7 @@
int subscribed; /* Is this call a subscription? */
int stateid;
+ int laststate; /* Last known state */
int dialogver;
struct ast_dsp *vad; /* Voice Activation Detection dsp */
@@ -601,6 +610,7 @@
char secret[80]; /* Password */
char md5secret[80]; /* Password in md5 */
char context[AST_MAX_CONTEXT]; /* Default context for incoming calls */
+ char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
char cid_num[80]; /* Caller ID num */
char cid_name[80]; /* Caller ID name */
char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
@@ -632,6 +642,7 @@
char md5secret[80]; /* Password in MD5 */
struct sip_auth *auth; /* Realm authentication list */
char context[AST_MAX_CONTEXT]; /* Default context for incoming calls */
+ char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
char username[80]; /* Temporary username until registration */
char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
int amaflags; /* AMA Flags (for billing) */
@@ -3797,7 +3808,12 @@
add_header(req, "From", ot);
add_header(req, "To", of);
}
- add_header(req, "Contact", p->our_contact);
+ if (sipmethod == SIP_MESSAGE) {
+ /* Add date header to MESSAGE */
+ append_date(req);
+ } else {
+ add_header(req, "Contact", p->our_contact);
+ }
copy_header(req, orig, "Call-ID");
add_header(req, "CSeq", tmp);
@@ -4465,101 +4481,144 @@
static int transmit_state_notify(struct sip_pvt *p, int state, int full)
{
char tmp[4000];
- int maxbytes = 0;
- int bytes = 0;
+ char *t = tmp;
+ int maxbytes = sizeof(tmp);
char from[256], to[256];
- char *t, *c, *a;
- char *mfrom, *mto;
+ char *mfrom, *mto, *c, *a;
struct sip_request req;
char clen[20];
+ char hint[AST_MAX_EXTENSION];
+ char *statestring = "terminated";
+ enum state {
+ NOTIFY_OPEN = 0,
+ NOTIFY_INUSE = 1,
+ NOTIFY_CLOSED = 2,
+ } local_state = NOTIFY_OPEN;
+ char *pidfstate = "--";
+ char *pidfnote= "Ready";
+
+ switch (state) {
+ case AST_EXTENSION_RINGING_AND_INUSE:
+ if (global_notifyringing)
+ statestring = "early";
+ else
+ statestring = "confirmed";
+ local_state = NOTIFY_INUSE;
+ pidfstate = "busy";
+ pidfnote = "Ringing";
+ break;
+ case AST_EXTENSION_RINGING:
+ statestring = "early";
+ local_state = NOTIFY_INUSE;
+ pidfstate = "busy";
+ pidfnote = "Ringing";
+ break;
+ case AST_EXTENSION_INUSE:
+ statestring = "confirmed";
+ local_state = NOTIFY_INUSE;
+ pidfstate = "busy";
+ pidfnote = "On the phone";
+ break;
+ case AST_EXTENSION_BUSY:
+ statestring = "confirmed";
+ local_state = NOTIFY_CLOSED;
+ pidfstate = "busy";
+ pidfnote = "On the phone";
+ break;
+ case AST_EXTENSION_UNAVAILABLE:
+ statestring = "confirmed";
+ local_state = NOTIFY_CLOSED;
+ pidfstate = "away";
+ pidfnote = "Unavailable";
+ break;
+ case AST_EXTENSION_NOT_INUSE:
+ default:
+ /* Default setting */
+ break;
+ }
+
+ /* Check which device/devices we are watching and if they are registered */
+ if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten)) {
+ /* If they are not registered, we will override notification and show no availability */
+ if (ast_device_state(hint) == AST_DEVICE_UNAVAILABLE) {
+ local_state = NOTIFY_CLOSED;
+ pidfstate = "away";
+ pidfnote = "Not online";
+ }
+ }
memset(from, 0, sizeof(from));
- memset(to, 0, sizeof(to));
ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from));
-
c = ditch_braces(from);
if (strncmp(c, "sip:", 4)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
return -1;
}
- if ((a = strchr(c, ';'))) {
+ if ((a = strchr(c, ';')))
*a = '\0';
- }
mfrom = c;
- reqprep(&req, p, SIP_NOTIFY, 0, 1);
-
- if (p->subscribed == 1) {
- ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
+ memset(to, 0, sizeof(to));
+ ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
+ c = ditch_braces(to);
+ if (strncmp(c, "sip:", 4)) {
+ ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
+ return -1;
+ }
+ if ((a = strchr(c, ';')))
+ *a = '\0';
+ mto = c;
- c = ditch_braces(to);
- if (strncmp(c, "sip:", 4)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
- return -1;
- }
- if ((a = strchr(c, ';'))) {
- *a = '\0';
- }
- mto = c;
+ reqprep(&req, p, SIP_NOTIFY, 0, 1);
+ switch (p->subscribed) {
+ case SUBSCRIBE_XPIDF_XML:
+ case SUBSCRIBE_CPIM_PIDF_XML:
add_header(&req, "Event", "presence");
add_header(&req, "Subscription-State", "active");
add_header(&req, "Content-Type", "application/xpidf+xml");
-
- if ((state==AST_EXTENSION_UNAVAILABLE) || (state==AST_EXTENSION_BUSY))
- state = 2;
- else if (state==AST_EXTENSION_INUSE)
- state = 1;
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "\n", mfrom);
+ ast_build_string(&t, &maxbytes, "\n", p->exten);
+ ast_build_string(&t, &maxbytes, "\n", mto);
+ ast_build_string(&t, &maxbytes, "\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed");
+ ast_build_string(&t, &maxbytes, "\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline");
+ ast_build_string(&t, &maxbytes, "\n\n\n");
+ break;
+ case SUBSCRIBE_PIDF_XML: /* Eyebeam supports this format */
+ add_header(&req, "Event", "presence");
+ add_header(&req, "Subscription-State", "active");
+ add_header(&req, "Content-Type", "application/pidf+xml");
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "\n", mfrom);
+ ast_build_string(&t, &maxbytes, "\n");
+ if (pidfstate[0] != '-')
+ ast_build_string(&t, &maxbytes, "%s\n", pidfstate);
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "%s\n", pidfnote); /* Note */
+ ast_build_string(&t, &maxbytes, "\n", p->exten); /* Tuple start */
+ ast_build_string(&t, &maxbytes, "%s\n", mto);
+ if (pidfstate[0] == 'b') /* Busy? Still open ... */
+ ast_build_string(&t, &maxbytes, "open\n");
else
- state = 0;
-
- t = tmp;
- maxbytes = sizeof(tmp);
- bytes = snprintf(t, maxbytes, "\n");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", mfrom);
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", p->exten);
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", mto);
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", !state ? "open" : (state==1) ? "inuse" : "closed");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", !state ? "online" : (state==1) ? "onthephone" : "offline");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n\n\n");
- } else {
+ ast_build_string(&t, &maxbytes, "%s\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed");
+ ast_build_string(&t, &maxbytes, "\n\n");
+ break;
+ case SUBSCRIBE_DIALOG_INFO_XML: /* SNOM subscribes in this format */
add_header(&req, "Event", "dialog");
add_header(&req, "Content-Type", "application/dialog-info+xml");
-
- t = tmp;
- maxbytes = sizeof(tmp);
- bytes = snprintf(t, maxbytes, "\n");
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n", p->dialogver++, full ? "full":"partial", mfrom);
- t += bytes;
- maxbytes -= bytes;
- bytes = snprintf(t, maxbytes, "\n\n");
+ ast_build_string(&t, &maxbytes, "\n");
+ ast_build_string(&t, &maxbytes, "\n", p->dialogver++, full ? "full":"partial", mto);
+ if (state == AST_EXTENSION_RINGING || (state==AST_EXTENSION_RINGING_AND_INUSE && global_notifyringing))
+ ast_build_string(&t, &maxbytes, "\n");
}
+
if (t > tmp + sizeof(tmp))
ast_log(LOG_WARNING, "Buffer overflow detected!! (Please file a bug report)\n");
@@ -5474,7 +5533,7 @@
#endif
)
return 0;
- if (sipmethod == SIP_REGISTER) {
+ if (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) {
/* On a REGISTER, we have to use 401 and its family of headers instead of 407 and its family
of headers -- GO SIP! Whoo hoo! Two things that do the same thing but are used in
different circumstances! What a surprise. */
@@ -5658,10 +5717,13 @@
return res;
}
-/*--- cb_extensionstate: Part of thte SUBSCRIBE support subsystem ---*/
+/*--- cb_extensionstate: Callback for the devicestate notification (SUBSCRIBE) support subsystem ---*/
+/* If you add an "hint" priority to the extension in the dial plan,
+ you will get notifications on device state changes */
static int cb_extensionstate(char *context, char* exten, int state, void *data)
{
struct sip_pvt *p = data;
+
if (state == -1) {
sip_scheddestroy(p, 15000);
p->stateid = -1;
@@ -5669,6 +5731,7 @@
}
transmit_state_notify(p, state, 1);
+ p->laststate = state;
if (option_debug > 1)
ast_verbose(VERBOSE_PREFIX_1 "Extension Changed %s new state %d for Notify User %s\n", exten, state, p->username);
@@ -6297,6 +6360,7 @@
ast_copy_string(p->cid_name, user->cid_name, sizeof(p->cid_name));
ast_copy_string(p->username, user->name, sizeof(p->username));
ast_copy_string(p->peersecret, user->secret, sizeof(p->peersecret));
+ ast_copy_string(p->subscribecontext, user->subscribecontext, sizeof(p->subscribecontext));
ast_copy_string(p->peermd5secret, user->md5secret, sizeof(p->peermd5secret));
ast_copy_string(p->accountcode, user->accountcode, sizeof(p->accountcode));
ast_copy_string(p->language, user->language, sizeof(p->language));
@@ -6359,6 +6423,7 @@
}
ast_copy_string(p->peersecret, peer->secret, sizeof(p->peersecret));
p->peersecret[sizeof(p->peersecret)-1] = '\0';
+ ast_copy_string(p->subscribecontext, peer->subscribecontext, sizeof(p->subscribecontext));
ast_copy_string(p->peermd5secret, peer->md5secret, sizeof(p->peermd5secret));
p->peermd5secret[sizeof(p->peermd5secret)-1] = '\0';
p->callingpres = peer->callingpres;
@@ -6465,16 +6530,28 @@
/*--- receive_message: Receive SIP MESSAGE method messages ---*/
-/* we handle messages within current calls currently */
+/* We only handle messages within current calls currently */
+/* Reference: RFC 3428 */
static void receive_message(struct sip_pvt *p, struct sip_request *req)
{
char buf[1024];
struct ast_frame f;
+ char *content_type;
+
+ content_type = get_header(req, "Content-Type");
+ if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
+ transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return;
+ }
if (get_msg_text(buf, sizeof(buf), req)) {
ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
+ transmit_response(p, "202 Accepted", req);
+ ast_set_flag(p, SIP_NEEDDESTROY);
return;
}
+
if (p->owner) {
if (sip_debug_test_pvt(p))
ast_verbose("Message received: '%s'\n", buf);
@@ -6485,7 +6562,13 @@
f.data = buf;
f.datalen = strlen(buf);
ast_queue_frame(p->owner, &f);
+ transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
+ } else { /* Message outside of a call, we do not support that */
+ ast_log(LOG_WARNING,"Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req,"To"), get_header(req,"From"), content_type, buf);
+ transmit_response(p, "405 Method Not Allowed", req); /* Good enough, or? */
}
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return;
}
/*--- sip_show_inuse: CLI Command to show calls within limits set by
@@ -7354,6 +7437,7 @@
static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions)
{
+#define FORMAT4 "%-15.15s %-10.10s %-21.21s %-15.15s %10s\n"
#define FORMAT3 "%-15.15s %-10.10s %-21.21s %-15.15s\n"
#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %s %s\n"
#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-6.6s%s %s\n"
@@ -7367,7 +7451,7 @@
if (!subscriptions)
ast_cli(fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Seq (Tx/Rx)", "Format", "Last Msg");
else
- ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "URI");
+ ast_cli(fd, FORMAT4, "Peer", "User", "Call ID", "Extension", "Last state");
while (cur) {
if (!cur->subscribed && !subscriptions) {
ast_cli(fd, FORMAT, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr),
@@ -7380,22 +7464,23 @@
numchans++;
}
if (cur->subscribed && subscriptions) {
- ast_cli(fd, FORMAT3, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr),
- ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username,
- cur->callid, cur->uri);
-
- }
+ ast_cli(fd, FORMAT4, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr),
+ ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username,
+ cur->callid, cur->exten, ast_state2str(cur->laststate));
+ numchans++;
+ }
cur = cur->next;
}
ast_mutex_unlock(&iflock);
if (!subscriptions)
- ast_cli(fd, "%d active SIP channel(s)\n", numchans);
+ ast_cli(fd, "%d active SIP channel%s\n", numchans, (numchans != 1) ? "s" : "");
else
- ast_cli(fd, "%d active SIP subscriptions(s)\n", numchans);
+ ast_cli(fd, "%d active SIP subscriptions%s\n", numchans, (numchans != 1) ? "s" : "");
return RESULT_SUCCESS;
#undef FORMAT
#undef FORMAT2
#undef FORMAT3
+#undef FORMAT4
}
/*--- complete_sipch: Support routine for 'sip show channel' CLI ---*/
@@ -8617,6 +8702,7 @@
by sending CANCEL */
ast_set_flag(p, SIP_PENDINGBYE);
p->authtries = 0;
+ ast_device_state_changed("SIP/%s", p->peername);
/* If I understand this right, the branch is different for a non-200 ACK only */
transmit_request(p, SIP_ACK, seqno, 0, 1);
check_pendings(p);
@@ -9377,16 +9463,12 @@
/*--- handle_request_message: Handle incoming MESSAGE request ---*/
static int handle_request_message(struct sip_pvt *p, struct sip_request *req, int debug, int ignore)
{
- if (p->lastinvite) {
- if (!ignore) {
- if (debug)
- ast_verbose("Receiving message!\n");
- receive_message(p, req);
- }
- transmit_response(p, "200 OK", req);
+ if (!ignore) {
+ if (debug)
+ ast_verbose("Receiving message!\n");
+ receive_message(p, req);
} else {
- transmit_response(p, "405 Method Not Allowed", req);
- ast_set_flag(p, SIP_NEEDDESTROY);
+ transmit_response(p, "202 Accepted", req);
}
return 1;
}
@@ -9422,7 +9504,9 @@
return 0;
}
/* Initialize the context if it hasn't been already */
- if (ast_strlen_zero(p->context))
+ if (p->subscribecontext && !ast_strlen_zero(p->subscribecontext))
+ ast_copy_string(p->context, p->subscribecontext, sizeof(p->context));
+ else if (ast_strlen_zero(p->context))
strcpy(p->context, default_context);
/* Get destination right away */
gotdest = get_destination(p, NULL);
@@ -9431,15 +9515,34 @@
if (gotdest < 0)
transmit_response(p, "404 Not Found", req);
else
- transmit_response(p, "484 Address Incomplete", req);
+ transmit_response(p, "484 Address Incomplete", req); /* Overlap dialing on SUBSCRIBE?? */
ast_set_flag(p, SIP_NEEDDESTROY);
} else {
+ char *event = get_header(req, "Event"); /* Get Event package name */
+ char *accept = get_header(req, "Accept");
+
/* Initialize tag */
p->tag = rand();
- if (!strcmp(get_header(req, "Accept"), "application/dialog-info+xml"))
- p->subscribed = 2;
- else if (!strcmp(get_header(req, "Accept"), "application/simple-message-summary")) {
- /* Looks like they actually want a mailbox */
+ if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
+ /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
+ if (strstr(accept, "application/dialog-info+xml"))
+ p->subscribed = SUBSCRIBE_DIALOG_INFO_XML;
+ /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
+ /* Should not be used for SUBSCRIBE, but anyway */
+ else if (!p->subscribed && strstr(accept, "application/cpim-pidf+xml"))
+ p->subscribed = SUBSCRIBE_CPIM_PIDF_XML; /* RFC 3863 format */
+ else if (!p->subscribed && strstr(accept, "application/pidf+xml"))
+ p->subscribed = SUBSCRIBE_PIDF_XML; /* RFC 3863 format */
+ else if (!p->subscribed && strstr(accept, "application/xpidf+xml"))
+ p->subscribed = SUBSCRIBE_XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
+ else {
+ /* Can't find a format for events that we know about */
+ transmit_response(p, "489 Bad Event", req);
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return 0;
+ }
+ } else if (!strcmp(event, "message-summary") && !strcmp(accept, "application/simple-message-summary")) {
+ /* Looks like they actually want a mailbox status */
/* At this point, we should check if they subscribe to a mailbox that
has the same extension as the peer or the mailbox id. If we configure
@@ -9450,9 +9553,8 @@
Since we do not act on this subscribe anyway, we might as well
accept any authenticated peer with a mailbox definition in their
- config section.
+ config section. */
- */
if (!ast_strlen_zero(mailbox)) {
found++;
}
@@ -9465,8 +9567,13 @@
ast_set_flag(p, SIP_NEEDDESTROY);
}
return 0;
- } else
- p->subscribed = 1;
+ } else { /* At this point, Asterisk does not understand the specified event */
+ transmit_response(p, "489 Bad Event", req);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event);
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return 0;
+ }
if (p->subscribed)
p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p);
}
@@ -9503,9 +9610,11 @@
ast_rtp_destroy(p->vrtp);
p->vrtp = NULL;
}
+ if (sipdebug || option_debug > 1)
+ ast_log(LOG_DEBUG, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username);
transmit_response(p, "200 OK", req);
- sip_scheddestroy(p, (p->expiry+10)*1000);
- transmit_state_notify(p, ast_extension_state(NULL, p->context, p->exten),1);
+ sip_scheddestroy(p, (p->expiry + 10) * 1000); /* Set timer for destruction of call at expiration */
+ transmit_state_notify(p, ast_extension_state(NULL, p->context, p->exten),1); /* Send first notification */
}
return 1;
}
@@ -10117,13 +10226,27 @@
if (p) {
found++;
res = AST_DEVICE_UNAVAILABLE;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Checking device state for peer %s\n", dest);
if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) &&
(!p->maxms || ((p->lastms > -1) && (p->lastms <= p->maxms)))) {
- /* peer found and valid */
- res = AST_DEVICE_UNKNOWN;
+ /* Peer is registred, or has default IP address and a valid registration */
+ /* Now check if we know anything about the state. The only way is by implementing
+ * call control with incominglimit=X in sip.conf where X > 0
+ * Check if the device has incominglimit, and if qualify=on, if the device
+ * is reachable */
+ if (p->incominglimit && (p->lastms == 0 || p->lastms <= p->maxms)) { /* Free for a call */
+ res = AST_DEVICE_NOT_INUSE;
+ if (p->inUse) /* On a call */
+ res = AST_DEVICE_BUSY;
+ } else { /* peer found and valid, state unknown */
+ res = AST_DEVICE_UNKNOWN;
+ }
}
}
if (!p && !found) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Checking device state for DNS host %s\n", dest);
hp = ast_gethostbyname(host, &ahp);
if (hp)
res = AST_DEVICE_UNKNOWN;
@@ -10453,6 +10576,8 @@
if (!strcasecmp(v->name, "context")) {
ast_copy_string(user->context, v->value, sizeof(user->context));
+ } else if (!strcasecmp(v->name, "subscribecontext")) {
+ ast_copy_string(user->subscribecontext, v->value, sizeof(user->subscribecontext));
} else if (!strcasecmp(v->name, "setvar")) {
varname = ast_strdupa(v->value);
if (varname && (varval = strchr(varname,'='))) {
@@ -10537,6 +10662,7 @@
SIP_DTMF | SIP_NAT | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE |
SIP_PROG_INBAND | SIP_OSPAUTH);
strcpy(peer->context, default_context);
+ strcpy(peer->subscribecontext, default_subscribecontext);
strcpy(peer->language, default_language);
strcpy(peer->musicclass, global_musicclass);
peer->addr.sin_port = htons(DEFAULT_SIP_PORT);
@@ -10578,9 +10704,9 @@
/* Already in the list, remove it and it will be added back (or FREE'd) */
found++;
} else {
- peer = malloc(sizeof(struct sip_peer));
+ peer = malloc(sizeof(*peer));
if (peer) {
- memset(peer, 0, sizeof(struct sip_peer));
+ memset(peer, 0, sizeof(*peer));
if (realtime)
rpeerobjs++;
else
@@ -10611,6 +10737,7 @@
peer->chanvars = NULL;
}
strcpy(peer->context, default_context);
+ strcpy(peer->subscribecontext, default_subscribecontext);
strcpy(peer->language, default_language);
strcpy(peer->musicclass, global_musicclass);
ast_copy_flags(peer, &global_flags, SIP_USEREQPHONE);
@@ -10660,6 +10787,8 @@
ast_callerid_split(v->value, peer->cid_name, sizeof(peer->cid_name), peer->cid_num, sizeof(peer->cid_num));
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(peer->context, v->value, sizeof(peer->context));
+ } else if (!strcasecmp(v->name, "subscribecontext")) {
+ ast_copy_string(peer->subscribecontext, v->value, sizeof(peer->subscribecontext));
} else if (!strcasecmp(v->name, "fromdomain"))
ast_copy_string(peer->fromdomain, v->value, sizeof(peer->fromdomain));
else if (!strcasecmp(v->name, "usereqphone"))
@@ -10853,6 +10982,7 @@
/* Initialize some reasonable defaults at SIP reload */
ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));
+ default_subscribecontext[0] = '\0';
default_language[0] = '\0';
default_fromdomain[0] = '\0';
default_qualify = 0;
@@ -10862,6 +10992,7 @@
sipdebug = 0;
ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent));
ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
+ global_notifyringing = 1;
ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm));
ast_copy_string(global_musicclass, "default", sizeof(global_musicclass));
ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
@@ -10951,6 +11082,8 @@
compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
+ } else if (!strcasecmp(v->name, "notifyringing")) {
+ global_notifyringing = ast_true(v->value);
} else if (!strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) {
ast_copy_string(global_musicclass, v->value, sizeof(global_musicclass));
} else if (!strcasecmp(v->name, "language")) {
Index: configs/sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.64
diff -u -r1.64 sip.conf.sample
--- configs/sip.conf.sample 9 Jun 2005 21:11:30 -0000 1.64
+++ configs/sip.conf.sample 25 Jul 2005 18:52:44 -0000
@@ -48,6 +48,9 @@
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
+;notifyringing=no ; Sent ringing (yes) or inuse (no) as state in
+ ; NOTIFY messages when observed extension is
+ ; in state RINGING_AND_INUSE (defaults to "yes")
;checkmwi=10 ; Default time between mailbox checks for peers
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
Index: include/asterisk/pbx.h
===================================================================
RCS file: /usr/cvsroot/asterisk/include/asterisk/pbx.h,v
retrieving revision 1.49
diff -u -r1.49 pbx.h
--- include/asterisk/pbx.h 15 Jul 2005 23:24:51 -0000 1.49
+++ include/asterisk/pbx.h 25 Jul 2005 18:52:53 -0000
@@ -42,6 +42,10 @@
#define AST_EXTENSION_BUSY 2
/*! All devices UNAVAILABLE/UNREGISTERED */
#define AST_EXTENSION_UNAVAILABLE 3
+/*! One or more devices are RINGING and none are INUSE */
+#define AST_EXTENSION_RINGING 4
+/*! One or more devices are RINGING and one or more are INUSE */
+#define AST_EXTENSION_RINGING_AND_INUSE 5
struct ast_context;
struct ast_exten;