Merged revisions 376998 via svnmerge from

file:///srv/subversion/repos/asterisk/trunk

........
  r376998 | oej | 2012-12-03 03:35:55 -0600 (Mon, 03 Dec 2012) | 4 lines
  
  Formatting changes
  
  Found a large amount of missing {} in the code before patching in another branch
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Automerge script
2012-12-03 10:20:59 +00:00
parent f598d0eb74
commit 129b1fba48
2 changed files with 59 additions and 31 deletions

View File

@@ -6078,41 +6078,49 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
}
}
}
if (!ast_strlen_zero(peer->fromuser))
if (!ast_strlen_zero(peer->fromuser)) {
ast_string_field_set(dialog, fromuser, peer->fromuser);
if (!ast_strlen_zero(peer->language))
}
if (!ast_strlen_zero(peer->language)) {
ast_string_field_set(dialog, language, peer->language);
}
/* Set timer T1 to RTT for this peer (if known by qualify=) */
/* Minimum is settable or default to 100 ms */
/* If there is a maxms and lastms from a qualify use that over a manual T1
value. Otherwise, use the peer's T1 value. */
if (peer->maxms && peer->lastms)
if (peer->maxms && peer->lastms) {
dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
else
} else {
dialog->timer_t1 = peer->timer_t1;
}
/* Set timer B to control transaction timeouts, the peer setting is the default and overrides
the known timer */
if (peer->timer_b)
if (peer->timer_b) {
dialog->timer_b = peer->timer_b;
else
} else {
dialog->timer_b = 64 * dialog->timer_t1;
}
if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
dialog->noncodeccapability |= AST_RTP_DTMF;
else
} else {
dialog->noncodeccapability &= ~AST_RTP_DTMF;
}
dialog->directmediaacl = ast_duplicate_acl_list(peer->directmediaacl);
if (peer->call_limit)
if (peer->call_limit) {
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
if (!dialog->portinuri)
}
if (!dialog->portinuri) {
dialog->portinuri = peer->portinuri;
}
dialog->chanvars = copy_vars(peer->chanvars);
if (peer->fromdomainport)
if (peer->fromdomainport) {
dialog->fromdomainport = peer->fromdomainport;
}
return 0;
}