Merged revisions 45262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45263 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -536,6 +536,8 @@ static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction schem
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
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static int noncodeccapability = AST_RTP_DTMF;
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static int global_ignoreoodreplies = 1;
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/* Object counters */
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static int suserobjs = 0; /*!< Static users */
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static int ruserobjs = 0; /*!< Realtime users */
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@ -4257,9 +4259,14 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si
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}
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}
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ast_mutex_unlock(&iflock);
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if (req->method == SIP_RESPONSE && global_ignoreoodreplies)
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return NULL;
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/* Allocate new call */
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if ((p = sip_alloc(callid, sin, 1, intended_method)))
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ast_mutex_lock(&p->lock);
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return p;
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}
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@ -9985,6 +9992,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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ast_cli(fd, " Allow subscriptions: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
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ast_cli(fd, " Allow overlap dialing: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
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ast_cli(fd, " Promsic. redir: %s\n", ast_test_flag(&global_flags[0], SIP_PROMISCREDIR) ? "Yes" : "No");
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ast_cli(fd, " Drop misc replies: %s\n", global_ignoreoodreplies ? "Yes" : "No");
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ast_cli(fd, " SIP domain support: %s\n", AST_LIST_EMPTY(&domain_list) ? "No" : "Yes");
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ast_cli(fd, " Call to non-local dom.: %s\n", allow_external_domains ? "Yes" : "No");
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ast_cli(fd, " URI user is phone no: %s\n", ast_test_flag(&global_flags[0], SIP_USEREQPHONE) ? "Yes" : "No");
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@ -14968,6 +14976,11 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
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} else if (!strcasecmp(v->name, "rfc2833compensate")) {
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ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
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ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
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} else if (!strcasecmp(v->name, "ignoreoodreplies")) {
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if (ast_true(v->value))
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global_ignoreoodreplies = 1;
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else
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global_ignoreoodreplies = 0;
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}
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return res;
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@ -15657,6 +15670,7 @@ static int reload_config(enum channelreloadreason reason)
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ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for peers, users: TRUE */
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ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP); /* Default for peers, users: TRUE */
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ast_set_flag(&global_flags[1], SIP_PAGE2_RTUPDATE);
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global_ignoreoodreplies = 1;
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/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
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ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));
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@ -132,6 +132,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; contrary to the RFC3551 specification, the peer _should_
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; be negotiating AAL2-G726-32 instead :-(
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;ignoreoodreplies = no ; If no then out of dialog replies will not be ignored
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;
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; If regcontext is specified, Asterisk will dynamically create and destroy a
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; NoOp priority 1 extension for a given peer who registers or unregisters with
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