Merged revisions 317283 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines
  
  Resolves a deadlock that occurs during sip_new
  
  This is based on an uncommitted patch by jpeeler for the issue.  Instead of
  relocking and then unlocking the channel though, we keep the lock on the channel
  until we are finished doing what we need to the channel.
  
  (closes issue #18441)
  Reported by: Alric
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -6722,7 +6722,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	sip_pvt_lock(i);
 	ast_channel_cc_params_init(tmp, i->cc_params);
 	tmp->caller.id.tag = ast_strdup(i->cid_tag);
-	ast_channel_unlock(tmp);
 
 	tmp->tech = ( ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ?  &sip_tech_info : &sip_tech;
 
@@ -6888,6 +6887,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 		pbx_builtin_setvar_helper(tmp, v->name, ast_get_encoded_str(v->value, valuebuf, sizeof(valuebuf)));
 	}
 
+	ast_channel_unlock(tmp); /* ast_hangup requires the channel to be unlocked */
+
 	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;