[asterisk-commits] mmichelson: branch 1.8 r375994 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Wed Nov  7 11:08:44 2012
New Revision: 375994

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375994
Log:
Remove some debugging that accidentally made it in the last commit.


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=375994&r1=375993&r2=375994
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Wed Nov  7 11:08:44 2012
@@ -5859,7 +5859,6 @@
 	const char *referer = NULL;   /* SIP referrer */
 	int cc_core_id;
 	char uri[SIPBUFSIZE] = "";
-	char capabilities[SIPBUFSIZE];
 
 	if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
 		ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
@@ -5966,8 +5965,6 @@
 	p->callingpres = ast_party_id_presentation(&ast->caller.id);
 	p->jointcapability = ast_rtp_instance_available_formats(p->rtp, p->capability, p->prefcodec);
 	p->jointnoncodeccapability = p->noncodeccapability;
-
-	ast_log(LOG_NOTICE, "jointcapability is %s\n", ast_getformatname_multiple(capabilities, SIPBUFSIZE, p->jointcapability));
 
 	/* If there are no audio formats left to offer, punt */
 	if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {