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Merged revisions 45262 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45263 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
file 2006-10-16 19:43:33 +00:00
parent 877564d66b
commit 7948364c8f
2 changed files with 16 additions and 0 deletions

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@ -536,6 +536,8 @@ 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_ignoreoodreplies = 1;
/* Object counters */
static int suserobjs = 0; /*!< Static users */
static int ruserobjs = 0; /*!< Realtime users */
@ -4257,9 +4259,14 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si
}
}
ast_mutex_unlock(&iflock);
if (req->method == SIP_RESPONSE && global_ignoreoodreplies)
return NULL;
/* Allocate new call */
if ((p = sip_alloc(callid, sin, 1, intended_method)))
ast_mutex_lock(&p->lock);
return p;
}
@ -9985,6 +9992,7 @@ 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 replies: %s\n", global_ignoreoodreplies ? "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");
@ -14968,6 +14976,11 @@ 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, "ignoreoodreplies")) {
if (ast_true(v->value))
global_ignoreoodreplies = 1;
else
global_ignoreoodreplies = 0;
}
return res;
@ -15657,6 +15670,7 @@ 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_ignoreoodreplies = 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));

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@ -132,6 +132,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
;ignoreoodreplies = no ; If no then out of dialog replies 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