- dynamically allocate the ast_jb structure that is on the channel structure

  so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
  from configuring a jitterbuffer on a new channel because of a memory
  allocation error
- On passing through these channel drivers, configure the jitterbuffer before
  starting the PBX thread instead of afterwards. If the pbx fails to start for
  whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
  possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
  NULL in failure conditions


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 13 insertions, 7 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 498087cdc..2dbf6ddd0 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3648,9 +3648,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	if (!ast_strlen_zero(i->musicclass))
 		ast_string_field_set(tmp, musicclass, i->musicclass);
 	i->owner = tmp;
-	ast_mutex_lock(&usecnt_lock);
-	usecnt++;
-	ast_mutex_unlock(&usecnt_lock);
 	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
 	ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
 	if (!ast_strlen_zero(i->cid_num)) 
@@ -3671,11 +3668,20 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	if (!ast_strlen_zero(i->callid))
 		pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
 	ast_setstate(tmp, state);
+	/* Configure the new channel jb */
+	if (i->rtp) {
+		if (ast_jb_configure(tmp, &global_jbconf)) {
+			ast_hangup(tmp);
+			i->owner = NULL;
+			return NULL;
+		}
+	}
 	if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
 		ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
 		tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
 		ast_hangup(tmp);
-		tmp = NULL;
+		i->owner = NULL;
+		return NULL;
 	}
 	/* Set channel variables for this call from configuration */
 	for (v = i->chanvars ; v ; v = v->next)
@@ -3684,9 +3690,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	if (recordhistory)
 		append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
 
-	/* Configure the new channel jb */
-	if (tmp && i && i->rtp)
-		ast_jb_configure(tmp, &global_jbconf);
+	ast_mutex_lock(&usecnt_lock);
+	usecnt++;
+	ast_mutex_unlock(&usecnt_lock);
 
 	return tmp;
 }