small formatting cleanup

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3d03892..66087f3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4500,6 +4500,8 @@ static int find_sdp(struct sip_request *req)
 
 /*! \brief Process SIP SDP offer, select formats and activate RTP channels
        If offer is rejected, we will not change any properties of the call
+       Return 0 on success, a negative value on errors.
+       Must be called after find_sdp().
 */
 static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 {
@@ -8712,6 +8714,16 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
        if (ast_strlen_zero(of))
                return AUTH_SUCCESSFUL;
 
+       if (1) {
+               const char *hdr = (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) ?
+                       "Authorization" : "Proxy-Authorization";
+               if ( (hdr = get_header(req, hdr)) && (hdr = strstr(hdr, "username=\"")) ) {
+                       ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
+                       of = from;
+                       of = strsep(&of, "\"");
+               }
+       }
+
        if (!authpeer)  /* If we are looking for a peer, don't check the user objects (or realtime) */
                user = find_user(of, 1);
 
@@ -11285,10 +11297,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
        /* 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 != 180) &&
-           (resp != 183))
+       if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
                resp = 183;
 
        switch (resp) {
@@ -11297,6 +11306,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                        sip_cancel_destroy(p);
                check_pendings(p);
                break;
+
        case 180:       /* 180 Ringing */
                if (!ast_test_flag(req, SIP_PKT_IGNORE))
                        sip_cancel_destroy(p);
@@ -11316,6 +11326,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                ast_set_flag(&p->flags[0], SIP_CAN_BYE);
                check_pendings(p);
                break;
+
        case 183:       /* Session progress */
                if (!ast_test_flag(req, SIP_PKT_IGNORE))
                        sip_cancel_destroy(p);
@@ -11330,6 +11341,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                ast_set_flag(&p->flags[0], SIP_CAN_BYE);
                check_pendings(p);
                break;
+
        case 200:       /* 200 OK on invite - someone's answering our call */
                if (!ast_test_flag(req, SIP_PKT_IGNORE))
                        sip_cancel_destroy(p);
@@ -11428,6 +11440,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                ast_set_flag(&p->flags[0], SIP_CAN_BYE);
                check_pendings(p);
                break;
+
        case 407: /* Proxy authentication */
        case 401: /* Www auth */
                /* First we ACK */
@@ -11449,6 +11462,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                        }
                }
                break;
+
        case 403: /* Forbidden */
                /* First we ACK */
                transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
@@ -11458,22 +11472,26 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
                ast_set_flag(&p->flags[0], SIP_ALREADYGONE);    
                break;
+
        case 404: /* Not found */
                transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
                if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
                ast_set_flag(&p->flags[0], SIP_ALREADYGONE);    
                break;
+
        case 481: /* Call leg does not exist */
                /* Could be REFER or INVITE */
                ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
                transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
                break;
+
        case 491: /* Pending */
                /* we have to wait a while, then retransmit */
                /* Transmission is rescheduled, so everything should be taken care of.
                        We should support the retry-after at some point */
                break;
+
        case 501: /* Not implemented */
                if (p->owner)
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);