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authorJoshua Colp <jcolp@digium.com>2006-10-16 20:26:56 +0000
committerJoshua Colp <jcolp@digium.com>2006-10-16 20:26:56 +0000
commitc62784c10dfc1fce5edd00fbfc616466a52d003e (patch)
treeac9d0018b11d81d7fabfe7e0f0eabf459a477f51
parentfe5641765e8a6607da5e9037feea4810b15b5ea8 (diff)
In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r--channels/chan_sip.c13
-rw-r--r--configs/sip.conf.sample2
2 files changed, 2 insertions, 13 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b9e57336f..8f9978cb3 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -536,8 +536,6 @@ 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 */
@@ -4260,8 +4258,8 @@ 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)
+ /* Responses can not create a pvt structure so drop it */
+ if (req->method == SIP_RESPONSE)
return NULL;
/* Allocate new call */
@@ -9993,7 +9991,6 @@ 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");
@@ -14977,11 +14974,6 @@ 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;
@@ -15671,7 +15663,6 @@ 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));
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index aa3512517..b6e84fa91 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -132,8 +132,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
-;ignoreoodresponses = no ; If no then out of dialog responses will not be ignored
-
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with