[svn-commits] mmichelson: branch 1.6.1 r201464 - in /branches/1.6.1: ./ channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: mmichelson
Date: Wed Jun 17 15:11:29 2009
New Revision: 201464

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=201464
Log:
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.
........

Modified:
    branches/1.6.1/   (props changed)
    branches/1.6.1/channels/chan_sip.c

Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=201464&r1=201463&r2=201464
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Wed Jun 17 15:11:29 2009
@@ -1345,7 +1345,7 @@
 	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 */
@@ -7046,7 +7046,7 @@
 	int last_rtpmap_codec=0;
 
 	char buf[SIPBUFSIZE];
-	uint64_t rua_version;
+	int64_t rua_version;
 	
 	int red_data_pt[10];
 	int red_num_gen = 0;
@@ -7122,7 +7122,7 @@
 		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;
 	}