Merged revisions 201462 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines
  
  Fix problem with no audio due to ignoring the SDP.
  
  A recent change to our SDP version comparison made audio not function
  on some calls. This was because of a test wherein we were trying to
  see if an unsigned value was less than 0. This is a dumb comparison
  and arguably the compiler should have warned about it. Alas, though,
  it slipped past. Now it's fixed by changing the variable to be a
  signed type.
  
  Found by several developers. Tested by mnicholson and dbrooks.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -1259,7 +1259,7 @@ struct sip_pvt {
 	char tag[11];				/*!< Our tag for this session */
 	int sessionid;				/*!< SDP Session ID */
 	int sessionversion;			/*!< SDP Session Version */
-	uint64_t sessionversion_remote;		/*!< Remote UA's SDP Session Version */
+	int64_t sessionversion_remote;		/*!< Remote UA's SDP Session Version */
 	int session_modify;			/*!< Session modification request true/false  */
 	struct sockaddr_in sa;			/*!< Our peer */
 	struct sockaddr_in redirip;		/*!< Where our RTP should be going if not to us */
@@ -6719,7 +6719,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	int last_rtpmap_codec=0;
 
 	char buf[SIPBUFSIZE];
-	uint64_t rua_version;
+	int64_t rua_version;
 
 	if (!p->rtp) {
 		ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
@@ -6788,7 +6788,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		ast_log(LOG_WARNING, "SDP sytax error in o= line\n");
 		return -1;
 	}
-	if (!sscanf(token, "%" SCNu64, &rua_version)) {
+	if (!sscanf(token, "%" SCNd64, &rua_version)) {
 		ast_log(LOG_WARNING, "SDP sytax error in o= line version\n");
 		return -1;
 	}