]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
fixes invite with replaces deadlock
authorDavid Vossel <dvossel@digium.com>
Tue, 23 Feb 2010 16:26:05 +0000 (16:26 +0000)
committerDavid Vossel <dvossel@digium.com>
Tue, 23 Feb 2010 16:26:05 +0000 (16:26 +0000)
(closes issue #16862)
Reported by: pwalker
Patches:
      replaces_deadlock_1.4 uploaded by dvossel (license 671)
Tested by: pwalker, dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 82516a21c14038c8bd76ba5f580822afd58e2b0d..e2db50d70c645dc63424a8ab823ac811e9d38c3e 100644 (file)
@@ -14763,6 +14763,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
        int gotdest;
        const char *p_replaces;
        char *replace_id = NULL;
+       int refer_locked = 0;
        const char *required;
        unsigned int required_profile = 0;
        struct ast_channel *c = NULL;           /* New channel */
@@ -14786,7 +14787,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        p->invitestate = INV_COMPLETED;
                        if (!p->lastinvite)
                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-                       return -1;
+                       res = -1;
+                       goto request_invite_cleanup;
                }
        }
 
@@ -14806,7 +14808,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        transmit_response(p, "482 Loop Detected", req);
                        p->invitestate = INV_COMPLETED;
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                } else {
                        /* This is a spiral. What we need to do is to just change the outgoing INVITE
                         * so that it now routes to the new Request URI. Since we created the INVITE ourselves
@@ -14831,7 +14834,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                         */
                        ast_string_field_set(p->owner, call_forward, peerorhost);
                        ast_queue_control(p->owner, AST_CONTROL_BUSY);
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                }
        }
        
@@ -14870,7 +14874,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        if (option_debug)
                                ast_log(LOG_DEBUG, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
                        /* Don't destroy dialog here */
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                }
        }
 
@@ -14888,7 +14893,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                ast_log(LOG_DEBUG, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
                        transmit_response_reliable(p, "400 Bad request", req);  /* The best way to not not accept the transfer */
                        /* Do not destroy existing call */
-                       return -1;
+                       res = -1;
+                       goto request_invite_cleanup;
                }
 
                if (sipdebug && option_debug > 2)
@@ -14902,7 +14908,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                        p->invitestate = INV_COMPLETED;
-                       return -1;
+                       res = -1;
+                       goto request_invite_cleanup;
                }
 
                /*  Todo: (When we find phones that support this)
@@ -14938,6 +14945,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
                        transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
                        error = 1;
+               } else {
+                       refer_locked = 1;
                }
 
                /* At this point, bot the pvt and the owner of the call to be replaced is locked */
@@ -14977,8 +14986,10 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                        ast_channel_unlock(p->refer->refer_call->owner);
                                }
                        }
+                       refer_locked = 0;
                        p->invitestate = INV_COMPLETED;
-                       return -1;
+                       res = -1;
+                       goto request_invite_cleanup;
                }
        }
 
@@ -15010,7 +15021,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                        transmit_response_reliable(p, "488 Not acceptable here", req);
                                        if (!p->lastinvite)
                                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-                                       return -1;
+                                       res = -1;
+                                       goto request_invite_cleanup;
                                }
                                ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
                        } else {
@@ -15040,7 +15052,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
                if (res == AUTH_CHALLENGE_SENT) {
                        p->invitestate = INV_COMPLETED;         /* Needs to restart in another INVITE transaction */
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                }
                if (res < 0) { /* Something failed in authentication */
                        if (res == AUTH_FAKE_AUTH) {
@@ -15053,7 +15066,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        p->invitestate = INV_COMPLETED; 
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                        ast_string_field_free(p, theirtag);
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
+
                }
 
                /* We have a succesful authentication, process the SDP portion if there is one */
@@ -15065,7 +15080,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                if (option_debug)
                                        ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
-                               return -1;
+                               res = -1;
+                               goto request_invite_cleanup;
                        }
                        if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
                                if (p->rtp) {
@@ -15102,7 +15118,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                p->invitestate = INV_COMPLETED; 
                        }
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                }
                gotdest = get_destination(p, NULL);     /* Get destination right away */
                get_rdnis(p, NULL);                     /* Get redirect information */
@@ -15129,7 +15146,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        p->invitestate = INV_COMPLETED; 
                        update_call_counter(p, DEC_CALL_LIMIT);
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-                       return 0;
+                       res = 0;
+                       goto request_invite_cleanup;
                } else {
                        /* If no extension was specified, use the s one */
                        /* Basically for calling to IP/Host name only */
@@ -15169,7 +15187,10 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                /* Go and take over the target call */
                if (sipdebug && option_debug > 3)
                        ast_log(LOG_DEBUG, "Sending this call to the invite/replcaes handler %s\n", p->callid);
-               return handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin, nounlock);
+
+               res = handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin, nounlock);
+               refer_locked = 0;
+               goto request_invite_cleanup;
        }
 
 
@@ -15357,6 +15378,17 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                }
        }
        return res;
+
+request_invite_cleanup:
+
+       if (refer_locked && p->refer && p->refer->refer_call) {
+               ast_mutex_unlock(&p->refer->refer_call->lock);
+               if (p->refer->refer_call->owner) {
+                       ast_channel_unlock(p->refer->refer_call->owner);
+               }
+       }
+
+       return res;
 }
 
 /*! \brief  Find all call legs and bridge transferee with target