diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2d380e9f17c6456e5a33618d6130cfe4821a9e94..76c8ae4d22d3481c1ac6acd5d1308d33b3636631 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -238,6 +238,21 @@ enum sip_result {
AST_FAILURE = -1,
};
+/*! \brief States for the INVITE transaction, not the dialog
+ \note this is for the INVITE that sets up the dialog
+*/
+enum invitestates {
+ INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
+ INV_CALLING, /*!< Invite sent, no answer */
+ INV_PROCEEDING, /*!< We got 1xx message */
+ INV_EARLY_MEDIA, /*!< We got 18x message with to-tag back */
+ INV_COMPLETED, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
+ INV_CONFIRMED, /*!< Confirmed response - we've got an ack (Incoming calls only) */
+ INV_TERMINATED, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
+ The only way out of this is a BYE from one side */
+ INV_CANCELLED /*!< Transaction cancelled by client or server in non-terminated state */
+};
+
/* Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
@@ -699,7 +714,7 @@ struct sip_auth {
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
-#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
+#define SIP_FREE_BIT (1 << 14) /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
@@ -870,6 +885,7 @@ struct sip_refer {
/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
struct sip_pvt {
ast_mutex_t pvt_lock; /*!< Dialog private lock */
+ enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
int method; /*!< SIP method that opened this dialog */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
@@ -2915,6 +2931,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
if (option_debug > 1)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
+ p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
@@ -3417,7 +3434,8 @@ static int sip_hangup(struct ast_channel *ast)
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE)) {
+ if (p->invitestate == INV_CALLING) {
+ /* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
@@ -7467,6 +7485,9 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
{
struct sip_request resp;
+ if (sipmethod == SIP_ACK)
+ p->invitestate = INV_CONFIRMED;
+
reqprep(&resp, p, sipmethod, seqno, newbranch);
add_header_contentLength(&resp, 0);
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
@@ -11646,7 +11667,7 @@ static void check_pendings(struct sip_pvt *p)
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
+ if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
@@ -11697,6 +11718,15 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
resp = 183;
+ /* Any response between 100 and 199 is PROCEEDING */
+ if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+ p->invitestate = INV_PROCEEDING;
+
+ /* Final response, not 200 ? */
+ if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
+ p->invitestate = INV_COMPLETED;
+
+
switch (resp) {
case 100: /* Trying */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
@@ -11714,13 +11744,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
}
}
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11729,13 +11759,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11833,8 +11863,8 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
+ p->invitestate = INV_TERMINATED;
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -13441,6 +13471,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result res;
@@ -13450,6 +13481,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
switch(res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "503 Unavailable", req);
else
@@ -13457,6 +13489,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "480 Temporarily Unavailable", req);
else
@@ -13493,6 +13526,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
+ p->invitestate = INV_COMPLETED;
ast_hangup(c);
sip_pvt_lock(p);
c = NULL;
@@ -13500,9 +13534,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
if (option_debug > 1)
@@ -13588,6 +13624,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
}
+ p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -13608,6 +13645,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@@ -14063,6 +14101,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@@ -14095,9 +14134,11 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
- if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
+ if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+ p->invitestate = INV_TERMINATED;
+
copy_request(&p->initreq, req);
if (sipdebug && option_debug)
ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
@@ -14667,6 +14708,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
+ p->invitestate = INV_CONFIRMED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {