[asterisk-commits] kmoore: branch 1.8 r413591 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: kmoore
Date: Fri May  9 17:56:14 2014
New Revision: 413591

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=413591
Log:
Fix 32bit build for chan_sip

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=413591&r1=413590&r2=413591
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Fri May  9 17:56:14 2014
@@ -11526,7 +11526,7 @@
 
 
 	if (debug)
-		ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", (long unsigned)codec, ast_getformatname(codec));
+		ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", (uint64_t)codec, ast_getformatname(codec));
 	if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
 		return;
 
@@ -11588,7 +11588,7 @@
 		return;
 
 	if (debug)
-		ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", (long unsigned)codec, ast_getformatname(codec));
+		ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", (uint64_t)codec, ast_getformatname(codec));
 
 	if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
 		return;