[Asterisk-cvs] asterisk/channels chan_h323.c,1.3,1.4
jeremy at lists.digium.com
jeremy at lists.digium.com
Mon Sep 22 01:18:52 CDT 2003
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv31123
Modified Files:
chan_h323.c
Log Message:
rollback transfer support...not properly implemented
Index: chan_h323.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_h323.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -d -r1.3 -r1.4
--- chan_h323.c 6 Sep 2003 20:29:25 -0000 1.3
+++ chan_h323.c 22 Sep 2003 06:20:01 -0000 1.4
@@ -232,6 +232,8 @@
strncpy(user->context, v->value, sizeof(user->context)-1);
} else if (!strcasecmp(v->name, "bridge")) {
user->bridge = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "nat")) {
+ user->nat = ast_true(v->value);
} else if (!strcasecmp(v->name, "noFastStart")) {
user->noFastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "noH245Tunneling")) {
@@ -497,6 +499,14 @@
/* Retrieve audio/etc from channel. Assumes p->lock is already held. */
struct ast_frame *f;
static struct ast_frame null_frame = { AST_FRAME_NULL, };
+
+ /* Only apply it for the first packet, we just need the correct ip/port */
+ if(p->nat)
+ {
+ ast_rtp_setnat(p->rtp,p->nat);
+ p->nat = 0;
+ }
+
f = ast_rtp_read(p->rtp);
/* Don't send RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && !(p->dtmfmode & H323_DTMF_RFC2833))
@@ -1031,6 +1041,7 @@
}
strncpy(p->context, user->context, sizeof(p->context)-1);
p->bridge = user->bridge;
+ p->nat = user->nat;
if (strlen(user->callerid))
strncpy(p->callerid, user->callerid, sizeof(p->callerid) - 1);
@@ -1056,9 +1067,7 @@
/* I know this is horrid, don't kill me saddam */
exit:
/* allocate a channel and tell asterisk about it */
- printf("exten b4: %s\n", p->exten);
c = oh323_new(p, AST_STATE_RINGING, cd.call_token);
-
if (!c) {
ast_log(LOG_ERROR, "Couldn't create channel. This is bad\n");
return 0;
@@ -1089,61 +1098,6 @@
}
#endif
-
-/* Call-back function that gets called on transfer
- *
- * Returns 1 on success
- */
-int setup_transfer_call(unsigned call_reference, const char *extension)
-{
- struct oh323_pvt *p;
- struct ast_channel *c = NULL;
- char exten[AST_MAX_EXTENSION];
- char *context;
-
- p = find_call(call_reference);
-
- if (!p) {
- ast_log(LOG_WARNING, "No such call %d.\n", call_reference);
- return -1;
- }
-
- if (!p->owner) {
- ast_log(LOG_WARNING, "Call %d has no owner.\n", call_reference);
- return -1;
- }
-
- memcpy(exten, extension, sizeof(exten));
-
- c = p->owner;
- if (c && c->bridge) {
- strncpy(exten, extension, sizeof(exten) - 1);
- context = strchr(exten, '@');
- if (context) {
- *context = '\0';
- context++;
- } else
- context = c->context;
- if (!strlen(context))
- context = c->bridge->context;
- if (ast_exists_extension(c->bridge, context, exten, 1, c->bridge->callerid)) {
-
- ast_log(LOG_NOTICE, "Transfering call %s to %s@%s.\n", c->bridge->name, exten, context);
-
- if (!ast_async_goto(c->bridge, context, exten, 1, 1))
- return 1;
-
- ast_log(LOG_WARNING, "Failed to transfer.\n");
- } else {
- ast_log(LOG_WARNING, "No such extension '%s' exists.\n", exten);
- }
- } else {
- ast_log(LOG_WARNING, "There is no call to transfer\n");
- }
- return 0;
-}
-
-
/**
* Call-back function that gets called for each rtp channel opened
*
@@ -1768,8 +1722,7 @@
/* Register our callback functions */
h323_callback_register(setup_incoming_call,
- setup_outgoing_call,
- setup_transfer_call,
+ setup_outgoing_call,
create_connection,
setup_rtp_connection,
cleanup_connection,
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