From: Russell Bryant <russell@russellbryant.com>
Date: Sun, 27 Feb 2005 02:37:23 +0000 (+0000)
Subject: move the processing of SDP data to after determining the user (bug #3660)
X-Git-Tag: 1.0.11.1~208
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=f1c93a8ff32a6721c12f9901ac087de84ee75524;p=thirdparty%2Fasterisk.git

move the processing of SDP data to after determining the user (bug #3660)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-0@5096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/CHANGES b/CHANGES
index 09e8040381..6c8f96a732 100755
--- a/CHANGES
+++ b/CHANGES
@@ -18,6 +18,9 @@ Asterisk 1.0.6
     -- Asterisk was not sending the same From: line in SIP messages during certain times.
        Fixed to make sure it stays the same. This makes some providers happier, to a working state.
     -- Certain circumstances involving a blank callerid caused asterisk to segmentation fault.
+    -- There was a problem incorrectly matching codec availablity when global preferences were 
+       different from that of the user.  To fix this, processing of SDP data has been moved 
+       to after determining who the call is coming from.
  -- chan_zap:
     -- During a certain scenario when using flash and '#' transfers you would hear the
        other person and the music they were hearing. This has been fixed.
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6873dce625..e0c94e6199 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7257,7 +7257,6 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 			/* We do NOT destroy p here, so that our response will be accepted */
 			return 0;
 		}
-		/* Process the SDP portion */
 		if (!ignore) {
 			/* Use this as the basis */
 			if (debug)
@@ -7269,16 +7268,6 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 			p->pendinginvite = seqno;
 			copy_request(&p->initreq, req);
 			check_via(p, req);
-			if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
-				if (process_sdp(p, req))
-					return -1;
-			} else {
-				p->jointcapability = p->capability;
-				ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
-			}
-			/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
-			if (p->owner)
-				ast_queue_frame(p->owner, &af);
 		} else if (debug)
 			ast_verbose("Ignoring this request\n");
 		if (!p->lastinvite && !ignore && !p->owner) {
@@ -7295,6 +7284,17 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 				}
 				return 0;
 			}
+			/* Process the SDP portion */
+			if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+				if (process_sdp(p, req))
+					return -1;
+			} else {
+				p->jointcapability = p->capability;
+				ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
+			}
+			/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
+			if (p->owner)
+				ast_queue_frame(p->owner, &af);
 			/* Initialize the context if it hasn't been already */
 			if (ast_strlen_zero(p->context))
 				strncpy(p->context, default_context, sizeof(p->context) - 1);