Small fixes to handle_request_invite

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7ef5e04..c9cb701 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10810,11 +10810,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                }
 
                /* We have a succesful authentication, process the SDP portion if there is one */
-               if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+               if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp") ) {
                        if (process_sdp(p, req)) {
                                /* Unacceptable codecs */
                                transmit_response_reliable(p, "488 Not acceptable here", req);
                                ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
+                               if (option_debug)
+                                       ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
                                return -1;
                        }
                } else {
@@ -10834,8 +10836,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                /* Check number of concurrent calls -vs- incoming limit HERE */
                if (option_debug)
                        ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username);
-               res = update_call_counter(p, INC_CALL_LIMIT);
-               if (res) {
+               if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
                        if (res < 0) {
                                ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
                                transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
@@ -10926,10 +10927,10 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                        ast_mutex_lock(&p->lock);
                                        c = NULL;
                                }
-                       } else {
+                       } else {        /* Pickup call in call group */
                                ast_mutex_unlock(&c->lock);
                                if (ast_pickup_call(c)) {
-                                       ast_log(LOG_NOTICE, "Nothing to pick up\n");
+                                       ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
                                        if (ignore)
                                                transmit_response(p, "503 Unavailable", req);
                                        else
@@ -10965,7 +10966,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        transmit_response(p, "100 Trying", req);
                }
        } else {
-               if (p && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY) && !ignore) {
+               if (p && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY)) {
                        if (!p->jointcapability) {
                                if (ignore)
                                        transmit_response(p, "488 Not Acceptable Here (codec error)", req);
@@ -10973,7 +10974,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                        transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req);
                                ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
                        } else {
-                               ast_log(LOG_NOTICE, "Unable to create/find channel\n");
+                               ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
                                if (ignore)
                                        transmit_response(p, "503 Unavailable", req);
                                else