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rollback transfer support...not properly implemented

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jeremy McNamara
2003-09-22 06:21:03 +00:00
parent 7c59caf852
commit a83662beda
3 changed files with 12 additions and 110 deletions
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71
channels/chan_h323.c
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@@ -232,6 +232,8 @@ static struct oh323_user *build_user(char *name, struct ast_variable *v)
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 @@ static struct ast_frame *oh323_rtp_read(struct oh323_pvt *p)
/* 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 @@ int setup_incoming_call(call_details_t cd)
}
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 @@ int setup_incoming_call(call_details_t cd)
/* 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 @@ if (!p) {
}
#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 @@ int load_module()
/* 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,