[asterisk-commits] file: trunk r45263 - in /trunk: ./
channels/chan_sip.c configs/sip.conf.sample
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Oct 16 12:43:34 MST 2006
Author: file
Date: Mon Oct 16 14:43:33 2006
New Revision: 45263
URL: http://svn.digium.com/view/asterisk?rev=45263&view=rev
Log:
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.
........
................
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=45263&r1=45262&r2=45263&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Oct 16 14:43:33 2006
@@ -535,6 +535,8 @@
/*! \brief Codecs that we support by default: */
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 */
@@ -4257,9 +4259,14 @@
}
}
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 @@
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 @@
} 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 @@
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));
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=45263&r1=45262&r2=45263&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Oct 16 14:43:33 2006
@@ -131,6 +131,8 @@
; for Sipura and Grandstream ATAs, among others). This is
; 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
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