handle_request_invite revise comment, fix coding guideline issues
authorDavid Vossel <dvossel@digium.com>
Fri, 19 Feb 2010 17:40:26 +0000 (17:40 +0000)
committerDavid Vossel <dvossel@digium.com>
Fri, 19 Feb 2010 17:40:26 +0000 (17:40 +0000)
I'm working with this code right now trying to analyze a deadlock.
This change is just to clean up a few things before I make a more
complex patch.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index aa3a706..086cec9 100644 (file)
@@ -18835,7 +18835,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                char exten[AST_MAX_EXTENSION];
                char context[AST_MAX_CONTEXT];
        } pickup = {
-               .exten = "",    
+               .exten = "",
        };
        st_ref = SESSION_TIMER_REFRESHER_AUTO;
 
@@ -18988,7 +18988,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 
                        If it's not in early mode, 486 Busy.
                */
-               
+
                /* Skip leading whitespace */
                replace_id = ast_skip_blanks(replace_id);
 
@@ -19036,14 +19036,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        }
                }
 
+               /* This locks both refer_call pvt and refer_call pvt's owner!!!*/
                if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
                        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;
                }
 
-               /* At this point, bot the pvt and the owner of the call to be replaced is locked */
-
                /* The matched call is the call from the transferer to Asterisk .
                        We want to bridge the bridged part of the call to the
                        incoming invite, thus taking over the refered call */
@@ -19136,7 +19135,6 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
        } else if (debug)
                ast_verbose("Ignoring this INVITE request\n");
 
-       
        if (!p->lastinvite && !req->ignore && !p->owner) {
                /* This is a new invite */
                /* Handle authentication if this is our first invite */
@@ -19155,7 +19153,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
                                transmit_response_reliable(p, "403 Forbidden", req);
                        }
-                       p->invitestate = INV_COMPLETED; 
+                       p->invitestate = INV_COMPLETED;
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                        ast_string_field_set(p, theirtag, NULL);
                        return 0;
@@ -19172,7 +19170,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        if (process_sdp(p, req, SDP_T38_INITIATE)) {
                                /* Unacceptable codecs */
                                transmit_response_reliable(p, "488 Not acceptable here", req);
-                               p->invitestate = INV_COMPLETED; 
+                               p->invitestate = INV_COMPLETED;
                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                ast_debug(1, "No compatible codecs for this SIP call.\n");
                                return -1;
@@ -19206,7 +19204,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
                                transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
                                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-                               p->invitestate = INV_COMPLETED; 
+                               p->invitestate = INV_COMPLETED;
                        }
                        return 0;
                }
@@ -19225,14 +19223,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                transmit_response_reliable(p, "484 Address Incomplete", req);
                        else {
                                char *decoded_exten = ast_strdupa(p->exten);
-                               
+
                                transmit_response_reliable(p, "404 Not Found", req);
                                ast_uri_decode(decoded_exten);
                                ast_log(LOG_NOTICE, "Call from '%s' to extension"
                                        " '%s' rejected because extension not found.\n",
                                        S_OR(p->username, p->peername), decoded_exten);
                        }
-                       p->invitestate = INV_COMPLETED; 
+                       p->invitestate = INV_COMPLETED;
                        update_call_counter(p, DEC_CALL_LIMIT);
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                        return 0;
@@ -19242,7 +19240,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        /* Basically for calling to IP/Host name only */
                        if (ast_strlen_zero(p->exten))
                                ast_string_field_set(p, exten, "s");
-                       /* Initialize our tag */        
+                       /* Initialize our tag */
 
                        make_our_tag(p->tag, sizeof(p->tag));
                        /* First invitation - create the channel */
@@ -19304,9 +19302,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                if (!ast_strlen_zero(p_uac_min_se)) {
                        rtn = parse_minse(p_uac_min_se, &uac_min_se);
                        if (rtn != 0) {
-                               transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
-                                       p->invitestate = INV_COMPLETED;
-                                       if (!p->lastinvite) {
+                               transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
+                               p->invitestate = INV_COMPLETED;
+                               if (!p->lastinvite) {
                                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                }
                                return -1;
@@ -19409,7 +19407,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
        if (!req->ignore && p)
                p->lastinvite = seqno;
 
-       if (replace_id) {       /* Attended transfer or call pickup - we're the target */
+       if (replace_id) {       /* Attended transfer or call pickup - we're the target */
                if (!ast_strlen_zero(pickup.exten)) {
                        append_history(p, "Xfer", "INVITE/Replace received");
 
@@ -19821,7 +19819,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                        pvt_set_needdestroy(p, "outside of dialog");
                }
                return 0;
-       }       
+       }
 
 
        /* Check if transfer is allowed from this device */
@@ -19834,7 +19832,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        }
 
        if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
-               /* Already have a pending REFER */      
+               /* Already have a pending REFER */
                transmit_response(p, "491 Request pending", req);
                append_history(p, "Xfer", "Refer failed. Request pending.");
                return 0;
@@ -19890,7 +19888,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                p->refer->localtransfer = 1;
        } else if (sipdebug)
                        ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
-       
+
        /* Is this a repeat of a current request? Ignore it */
        /* Don't know what else to do right now. */
        if (req->ignore)
@@ -19949,7 +19947,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
        }
 
-       ast_set_flag(&p->flags[0], SIP_GOTREFER);       
+       ast_set_flag(&p->flags[0], SIP_GOTREFER);
 
        /* Attended transfer: Find all call legs and bridge transferee with target*/
        if (p->refer->attendedtransfer) {
@@ -19968,7 +19966,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                *nounlock = 1;
                ast_channel_unlock(current.chan1);
                copy_request(&current.req, req);
-               ast_clear_flag(&p->flags[0], SIP_GOTREFER);     
+               ast_clear_flag(&p->flags[0], SIP_GOTREFER);
                p->refer->status = REFER_200OK;
                append_history(p, "Xfer", "REFER to call parking.");
                ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans, "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransfer2Parking: Yes\r\n",
@@ -20022,7 +20020,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        /* FAKE ringing if not attended transfer */
        if (!p->refer->attendedtransfer)
                transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE);
-               
+
        /* For blind transfer, this will lead to a new call */
        /* For attended transfer to remote host, this will lead to
           a new SIP call with a replaces header, if the dial plan allows it