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Merged revisions 317283 via svnmerge from

Browse Source 
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jonathan Rose
2011-05-05 19:33:11 +00:00
parent 4d612d126b
commit 932e34ee62
1 changed files with 2 additions and 1 deletions
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3
channels/chan_sip.c
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@@ -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;