remove the option that was not needed, really.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5071d7b9196b2f81ec01b159e7d7ebb8ee59e74f..a5d1d4d6fd06dec6a33f2ff9ef9974ba4fde0d3a 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -541,8 +541,6 @@ static enum transfermodes global_allowtransfer;     /*!< SIP Refer restriction schem
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
 static int noncodeccapability = AST_RTP_DTMF;
 
-static int global_ignoreoodresponses = 1;
-
 /* Object counters */
 static int suserobjs = 0;                /*!< Static users */
 static int ruserobjs = 0;                /*!< Realtime users */
@@ -4270,7 +4268,7 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si
        ast_mutex_unlock(&iflock);
 
        /* If this is a response and we have ignoring of out of dialog responses turned on, then drop it */
-       if (req->method == SIP_RESPONSE && global_ignoreoodresponses)
+       if (req->method == SIP_RESPONSE)
                return NULL;
 
        /* Allocate new call */
@@ -10014,7 +10012,6 @@ static int sip_show_settings(int fd, int argc, char *argv[])
        ast_cli(fd, "  Allow subscriptions:    %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
        ast_cli(fd, "  Allow overlap dialing:  %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
        ast_cli(fd, "  Promsic. redir:         %s\n", ast_test_flag(&global_flags[0], SIP_PROMISCREDIR) ? "Yes" : "No");
-       ast_cli(fd, "  Drop misc responses:    %s\n", global_ignoreoodresponses ? "Yes" : "No");
        ast_cli(fd, "  SIP domain support:     %s\n", AST_LIST_EMPTY(&domain_list) ? "No" : "Yes");
        ast_cli(fd, "  Call to non-local dom.: %s\n", allow_external_domains ? "Yes" : "No");
        ast_cli(fd, "  URI user is phone no:   %s\n", ast_test_flag(&global_flags[0], SIP_USEREQPHONE) ? "Yes" : "No");
@@ -15034,12 +15031,7 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
        } else if (!strcasecmp(v->name, "rfc2833compensate")) {
                ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
                ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
-       } else if (!strcasecmp(v->name, "ignoreoodresponses")) {
-               if (ast_true(v->value))
-                       global_ignoreoodresponses = 1;
-               else
-                       global_ignoreoodresponses = 0;
-       }
+       } 
 
        return res;
 }
@@ -15717,7 +15709,6 @@ static int reload_config(enum channelreloadreason reason)
        ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);       /* Default for peers, users: TRUE */
        ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP);         /* Default for peers, users: TRUE */
        ast_set_flag(&global_flags[1], SIP_PAGE2_RTUPDATE);
-       global_ignoreoodresponses = 1;
 
        /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
        ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));