[asterisk-commits] russell: branch 1.2 r61376 -
/branches/1.2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Wed Apr 11 07:02:54 MST 2007
Author: russell
Date: Wed Apr 11 09:02:54 2007
New Revision: 61376
URL: http://svn.digium.com/view/asterisk?view=rev&rev=61376
Log:
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
Modified:
branches/1.2/channels/chan_sip.c
Modified: branches/1.2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?view=diff&rev=61376&r1=61375&r2=61376
==============================================================================
--- branches/1.2/channels/chan_sip.c (original)
+++ branches/1.2/channels/chan_sip.c Wed Apr 11 09:02:54 2007
@@ -12630,8 +12630,7 @@
ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
peer->maxms = 0;
}
- } else if (strcasecmp(v->name, "type"))
- ast_log(LOG_WARNING, "Ignoring unknown option '%s' at line %d of sip.conf!\n", v->name, v->lineno);
+ }
v = v->next;
}
if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(&peer->flags_page2, SIP_PAGE2_DYNAMIC) && realtime) {
@@ -12927,9 +12926,8 @@
}
} else if (!strcasecmp(v->name, "callevents")) {
callevents = ast_true(v->value);
- } else
- ast_log(LOG_WARNING, "Ignoring unknown option '%s' at line %d of sip.conf!\n", v->name, v->lineno);
- v = v->next;
+ }
+ v = v->next;
}
if (!allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
@@ -12943,8 +12941,7 @@
/* Format for authentication is auth = username:password at realm */
if (!strcasecmp(v->name, "auth")) {
authl = add_realm_authentication(authl, v->value, v->lineno);
- } else
- ast_log(LOG_WARNING, "Ignoring unknown option '%s' at line %d of sip.conf!\n", v->name, v->lineno);
+ }
v = v->next;
}
More information about the asterisk-commits
mailing list