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Remove some debugging that accidentally made it in the last commit.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2012-11-07 17:08:44 +00:00
parent e95efa6c50
commit a63451b81c
1 changed files with 0 additions and 3 deletions
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channels/chan_sip.c
-3
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@@ -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);