From a63451b81caf333c249ec369ecf0bee65ca16fd0 Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Wed, 7 Nov 2012 17:08:44 +0000 Subject: [PATCH] Remove some debugging that accidentally made it in the last commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1a2073f5cd..5f68ff2d16 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5859,7 +5859,6 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) 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); @@ -5967,8 +5966,6 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) 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)) { ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); -- 2.47.3