Formatting changes only

--Denna och nedanstående rader kommer inte med i loggmeddelandet--

M    channels/chan_sip.c

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a35c02f..ce321e1 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -19921,10 +19921,11 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        struct ast_party_connected_line connected;
        struct ast_set_party_connected_line update_connected;
 
-       if (reinvite)
+       if (reinvite) {
                ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
-       else
+       } else {
                ast_debug(4, "SIP response %d to standard invite\n", resp);
+       }
 
        if (p->alreadygone) { /* This call is already gone */
                ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
@@ -19938,8 +19939,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        /* RFC3261 says we must treat every 1xx response (but not 100)
           that we don't recognize as if it was 183.
        */
-       if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
+       if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183) {
                resp = 183;
+       }
 
        /* For INVITE, treat all 2XX responses as we would a 200 response */
        if ((resp >= 200) && (resp < 300)) {
@@ -19947,16 +19949,19 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        }
 
        /* Any response between 100 and 199 is PROCEEDING */
-       if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+       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 ))
+       if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) {
                p->invitestate = INV_COMPLETED;
+       }
        
        /* Final response, clear out pending invite */
-       if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
+       if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
                p->pendinginvite = 0;
+       }
 
        /* If this is a response to our initial INVITE, we need to set what we can use
         * for this peer.
@@ -19968,15 +19973,17 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        switch (resp) {
        case 100:       /* Trying */
        case 101:       /* Dialog establishment */
-               if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
+               if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
                        ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+               }
                check_pendings(p);
                break;
 
        case 180:       /* 180 Ringing */
        case 182:       /* 182 Queued */
-               if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
+               if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
                        ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+               }
                if (!req->ignore && p->owner) {
                        if (get_rpid(p, req)) {
                                /* Queue a connected line update */
@@ -20005,8 +20012,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                        }
                }
                if (find_sdp(req)) {
-                       if (p->invitestate != INV_CANCELLED)
+                       if (p->invitestate != INV_CANCELLED) {
                                p->invitestate = INV_EARLY_MEDIA;
+                       }
                        res = process_sdp(p, req, SDP_T38_NONE);
                        if (!req->ignore && p->owner) {
                                /* Queue a progress frame only if we have SDP in 180 or 182 */
@@ -20037,8 +20045,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                break;
 
        case 183:       /* Session progress */
-               if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
+               if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
                        ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+               }
                if (!req->ignore && p->owner) {
                        if (get_rpid(p, req)) {
                                /* Queue a connected line update */
@@ -20063,8 +20072,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                        sip_handle_cc(p, req, AST_CC_CCNR);
                }
                if (find_sdp(req)) {
-                       if (p->invitestate != INV_CANCELLED)
+                       if (p->invitestate != INV_CANCELLED) {
                                p->invitestate = INV_EARLY_MEDIA;
+                       }
                        res = process_sdp(p, req, SDP_T38_NONE);
                        if (!req->ignore && p->owner) {
                                /* Queue a progress frame */
@@ -20084,8 +20094,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                break;
 
        case 200:       /* 200 OK on invite - someone's answering our call */
-               if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
+               if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
                        ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+               }
                p->authtries = 0;
                if (find_sdp(req)) {
                        if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
@@ -20136,14 +20147,16 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                        update_call_counter(p, DEC_CALL_RINGING);
                        parse_ok_contact(p, req);
                        /* Save Record-Route for any later requests we make on this dialogue */
-                       if (!reinvite)
+                       if (!reinvite) {
                                build_route(p, req, 1);
+                       }
 
                        if(set_address_from_contact(p)) {
                                /* Bad contact - we don't know how to reach this device */
                                /* We need to ACK, but then send a bye */
-                               if (!p->route && !req->ignore)
+                               if (!p->route && !req->ignore) {
                                        ast_set_flag(&p->flags[0], SIP_PENDINGBYE);     
+                               }
                        }
 
                }
@@ -20151,10 +20164,11 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                if (!req->ignore && p->owner) {
                        if (!reinvite) {
                                ast_queue_control(p->owner, AST_CONTROL_ANSWER);
-                               if (sip_cfg.callevents)
+                               if (sip_cfg.callevents) {
                                        manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
                                                "Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
                                                p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
+                               }
                        } else {        /* RE-invite */
                                if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
                                        ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
@@ -20166,8 +20180,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                         /* It's possible we're getting an 200 OK after we've tried to disconnect
                                  by sending CANCEL */
                        /* First send ACK, then send bye */
-                       if (!req->ignore)
+                       if (!req->ignore) {
                                ast_set_flag(&p->flags[0], SIP_PENDINGBYE);     
+                       }
                }
 
                /* Check for Session-Timers related headers */
@@ -20213,20 +20228,23 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        case 401: /* Www auth */
                /* First we ACK */
                xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-               if (p->options)
+               if (p->options) {
                        p->options->auth_type = resp;
+               }
 
                /* Then we AUTH */
                ast_string_field_set(p, theirtag, NULL);        /* forget their old tag, so we don't match tags when getting response */
                if (!req->ignore) {
-                       if (p->authtries < MAX_AUTHTRIES)
+                       if (p->authtries < MAX_AUTHTRIES) {
                                p->invitestate = INV_CALLING;
+                       }
                        if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
                                ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", sip_get_header(&p->initreq, "From"));
                                pvt_set_needdestroy(p, "failed to authenticate on INVITE");
                                sip_alreadygone(p);
-                               if (p->owner)
+                               if (p->owner) {
                                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+                               }
                        }
                }
                break;
@@ -20257,8 +20275,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                /* Could be REFER caused INVITE with replaces */
                ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
                xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-               if (p->owner)
+               if (p->owner) {
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+               }
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                break;
 
@@ -20272,8 +20291,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
                append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
                ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
-               if (p->owner)
+               if (p->owner) {
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+               }
                break;
 
                
@@ -20300,18 +20320,21 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
                        change_t38_state(p, T38_REJECTED);
                        /* Try to reset RTP timers */
+                       /* XXX Why is this commented away??? */
                        //ast_rtp_set_rtptimers_onhold(p->rtp);
 
                        /* Trigger a reinvite back to audio */
                        transmit_reinvite_with_sdp(p, FALSE, FALSE);
                } else {
                        /* We can't set up this call, so give up */
-                       if (p->owner && !req->ignore)
+                       if (p->owner && !req->ignore) {
                                ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+                       }
                        pvt_set_needdestroy(p, "received 488 response");
                        /* If there's no dialog to end, then mark p as already gone */
-                       if (!reinvite)
+                       if (!reinvite) {
                                sip_alreadygone(p);
+                       }
                }
                break;
        case 491: /* Pending */
@@ -20342,12 +20365,14 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
        case 405: /* Not allowed */
        case 501: /* Not implemented */
                xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-               if (p->owner)
+               if (p->owner) {
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+               }
                break;
        }
-       if (xmitres == XMIT_ERROR)
+       if (xmitres == XMIT_ERROR) {
                ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
+       }
 }
 
 /* \brief Handle SIP response in NOTIFY transaction