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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1a2073f5cd84f25a89affe0328fe175be6eb2ba9..5f68ff2d163481639dadacaf6528bfc29298aae3 100644 (file)
--- 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);