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authorjeremy <jeremy@f38db490-d61c-443f-a65b-d21fe96a405b>2003-12-24 22:38:24 +0000
committerjeremy <jeremy@f38db490-d61c-443f-a65b-d21fe96a405b>2003-12-24 22:38:24 +0000
commit73f7a57e9937a37f4357e6546ebbc870d25acf7b (patch)
tree12fff20e8cc06ef58f7d774b236ab51cc51dc9ef /channels/chan_h323.c
parent5c45c7bedd94300a8403ebcb091d4db972aaa20c (diff)
totally revert that highly broken patch. Please test your code before submitting diffs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1881 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_h323.c')
-rwxr-xr-xchannels/chan_h323.c410
1 files changed, 70 insertions, 340 deletions
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index eeeb001ea..15e4b5b22 100755
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -46,7 +46,6 @@
#include <asterisk/callerid.h>
#include <asterisk/cli.h>
#include <asterisk/dsp.h>
-#include <asterisk/translate.h>
#include <sys/socket.h>
#include <net/if.h>
#include <errno.h>
@@ -60,19 +59,6 @@
#include "h323/chan_h323.h"
-#define TRUE 1
-#define FALSE 0
-
-/* from rtp.c to translate RTP's payload type to Asterisk's frame subcode */
-struct rtpPayloadType {
- int isAstFormat; // whether the following code is an AST_FORMAT
- int code;
-};
-
-call_options_t global_options;
-
-static struct sockaddr_in bindaddr;
-
/** String variables required by ASTERISK */
static char *type = "H323";
static char *desc = "The NuFone Network's Open H.323 Channel Driver";
@@ -120,8 +106,6 @@ struct oh323_pvt {
int bridge; /* Determine of we should native bridge or not*/
char exten[AST_MAX_EXTENSION]; /* Requested extension */
char context[AST_MAX_EXTENSION]; /* Context where to start */
- char dnid[AST_MAX_EXTENSION]; /* Called number */
- char rdnis[AST_MAX_EXTENSION]; /* Redirecting number */
char username[81]; /* H.323 alias using this channel */
char accountcode[256]; /* Account code */
int amaflags; /* AMA Flags */
@@ -261,37 +245,17 @@ static struct oh323_user *build_user(char *name, struct ast_variable *v)
} else if (!strcasecmp(v->name, "nat")) {
user->nat = ast_true(v->value);
} else if (!strcasecmp(v->name, "noFastStart")) {
- user->call_options.noFastStart = ast_true(v->value);
+ user->noFastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "noH245Tunneling")) {
- user->call_options.noH245Tunnelling = ast_true(v->value);
+ user->noH245Tunneling = ast_true(v->value);
} else if (!strcasecmp(v->name, "noSilenceSuppression")) {
- user->call_options.noSilenceSuppression = ast_true(v->value);
+ user->noSilenceSuppression = ast_true(v->value);
} else if (!strcasecmp(v->name, "secret")) {
strncpy(user->secret, v->value, sizeof(user->secret)-1);
} else if (!strcasecmp(v->name, "callerid")) {
strncpy(user->callerid, v->value, sizeof(user->callerid)-1);
} else if (!strcasecmp(v->name, "accountcode")) {
strncpy(user->accountcode, v->value, sizeof(user->accountcode)-1);
- } else if (!strcasecmp(v->name, "progress_setup")) {
- int progress_setup = atoi(v->value);
- if((progress_setup != 0) &&
- (progress_setup != 1) &&
- (progress_setup != 3) &&
- (progress_setup != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_setup at line %d, assuming 0\n", progress_setup, v->lineno);
- progress_setup = 0;
- }
- user->call_options.progress_setup = progress_setup;
- } else if (!strcasecmp(v->name, "progress_alert")) {
- int progress_alert = atoi(v->value);
- if((progress_alert != 0) &&
- (progress_alert != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_alert at line %d, assuming 0\n", progress_alert, v->lineno);
- progress_alert = 0;
- }
- user->call_options.progress_alert = progress_alert;
- } else if (!strcasecmp(v->name, "progress_audio")) {
- user->call_options.progress_audio = ast_true(v->value);
} else if (!strcasecmp(v->name, "incominglimit")) {
user->incominglimit = atoi(v->value);
if (user->incominglimit < 0)
@@ -368,31 +332,11 @@ static struct oh323_peer *build_peer(char *name, struct ast_variable *v)
} else if (!strcasecmp(v->name, "bridge")) {
peer->bridge = ast_true(v->value);
} else if (!strcasecmp(v->name, "noFastStart")) {
- peer->call_options.noFastStart = ast_true(v->value);
+ peer->noFastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "noH245Tunneling")) {
- peer->call_options.noH245Tunnelling = ast_true(v->value);
+ peer->noH245Tunneling = ast_true(v->value);
} else if (!strcasecmp(v->name, "noSilenceSuppression")) {
- peer->call_options.noSilenceSuppression = ast_true(v->value);
- } else if (!strcasecmp(v->name, "progress_setup")) {
- int progress_setup = atoi(v->value);
- if((progress_setup != 0) &&
- (progress_setup != 1) &&
- (progress_setup != 3) &&
- (progress_setup != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_setup at line %d, assuming 0\n", progress_setup, v->lineno);
- progress_setup = 0;
- }
- peer->call_options.progress_setup = progress_setup;
- } else if (!strcasecmp(v->name, "progress_alert")) {
- int progress_alert = atoi(v->value);
- if((progress_alert != 0) &&
- (progress_alert != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_alert at line %d, assuming 0\n", progress_alert, v->lineno);
- progress_alert = 0;
- }
- peer->call_options.progress_alert = progress_alert;
- } else if (!strcasecmp(v->name, "progress_audio")) {
- peer->call_options.progress_audio = ast_true(v->value);
+ peer->noSilenceSuppression = ast_true(v->value);
} else if (!strcasecmp(v->name, "outgoinglimit")) {
peer->outgoinglimit = atoi(v->value);
if (peer->outgoinglimit > 0)
@@ -483,16 +427,16 @@ static int oh323_call(struct ast_channel *c, char *dest, int timeout)
} else {
p->calloptions.callerid = strdup(c->callerid);
}
- }
+ }
- res = h323_make_call(called_addr, &(p->cd), &p->calloptions);
+ res = h323_make_call(called_addr, &(p->cd), p->calloptions);
if (res) {
ast_log(LOG_NOTICE, "h323_make_call failed(%s)\n", c->name);
return -1;
}
- ast_setstate(c, AST_STATE_RING);
+ ast_setstate(c, AST_STATE_RINGING);
return 0;
}
@@ -560,31 +504,20 @@ static int oh323_hangup(struct ast_channel *c)
return 0;
}
-/* Pass channel struct too to allow RTCP handling */
-static struct ast_frame *oh323_rtp_read(struct ast_channel *c, struct oh323_pvt *p)
+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;
- }
-
- switch(c->fdno) {
- case 0: /* RTP stream */
- f = ast_rtp_read(p->rtp);
- break;
- case 1: /* RTCP stream */
- f = ast_rtcp_read(p->rtp);
- break;
- default:
- f = &null_frame;
- }
+ /* 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))
return &null_frame;
@@ -592,12 +525,10 @@ static struct ast_frame *oh323_rtp_read(struct ast_channel *c, struct oh323_pvt
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
if (f->subclass != p->owner->nativeformats) {
- /* Must be handled on opening logical channel */
ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
p->owner->nativeformats = f->subclass;
ast_set_read_format(p->owner, p->owner->readformat);
- /* Don't set write format because it will be set up when channel started */
-// ast_set_write_format(p->owner, p->owner->writeformat);
+ ast_set_write_format(p->owner, p->owner->writeformat);
}
/* Do in-band DTMF detection */
@@ -619,8 +550,7 @@ static struct ast_frame *oh323_read(struct ast_channel *c)
struct ast_frame *fr;
struct oh323_pvt *p = c->pvt->pvt;
ast_mutex_lock(&p->lock);
- /* Pass channel structure to handle other streams than just RTP */
- fr = oh323_rtp_read(c, p);
+ fr = oh323_rtp_read(p);
ast_mutex_unlock(&p->lock);
return fr;
}
@@ -629,7 +559,6 @@ static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
{
struct oh323_pvt *p = c->pvt->pvt;
int res = 0;
- int need_frfree = 0; /* Does ast_frfree() call required? */
if (frame->frametype != AST_FRAME_VOICE) {
if (frame->frametype == AST_FRAME_IMAGE)
return 0;
@@ -639,47 +568,9 @@ static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
}
} else {
if (!(frame->subclass & c->nativeformats)) {
- if(!(frame->subclass & c->writeformat)) { /* Someone sent frame with old format */
- ast_log(LOG_WARNING, "Asked to transmit frame type %s from %s by '%s', while native formats is %s (read/write = %s/%s)\n",
- ast_getformatname(frame->subclass), frame->src, c->name, ast_getformatname(c->nativeformats), ast_getformatname(c->readformat), ast_getformatname(c->writeformat));
- return (c->nativeformats ? 0 : -1);
- } else {
- /* Frame goes from RTP is not in our native
- * format - try to translate it... Or we must
- * just drop it?
- */
-
- /* Sometimes translation table isn't set
- * correctly but writeformat is invalid,
- * so force required translation allocation
- */
- ast_set_write_format(c, c->nativeformats);
- ast_set_write_format(c, frame->subclass);
-
- /* Translate it on-the-fly */
- if (c->pvt->writetrans) {
- struct ast_frame *frame1;
- ast_log(LOG_WARNING, "Asked to transmit frame type %s from %s by '%s', while native formats is %s (read/write = %s/%s) - 2 TRANSLATE\n",
- ast_getformatname(frame->subclass), frame->src, c->name, ast_getformatname(c->nativeformats), ast_getformatname(c->readformat), ast_getformatname(c->writeformat));
- /* Allocate new frame with translated context.
- * Don't free frame because it will be freed on
- * upper layer (RTP).
- */
- frame1 = ast_translate(c->pvt->writetrans, frame, 0);
- if(frame1) {
- /* Substitute passed frame with translated and
- mark it for freeing before return */
- frame = frame1;
- need_frfree = 1;
- }
- else
- ast_log(LOG_WARNING, "Unable to translate frame type %s to %s\n", ast_getformatname(frame->subclass), ast_getformatname(c->nativeformats));
- } else {
- ast_log(LOG_WARNING, "Asked to transmit frame type %s from %s by '%s', while native formats is %s (read/write = %s/%s)\n",
- ast_getformatname(frame->subclass), frame->src, c->name, ast_getformatname(c->nativeformats), ast_getformatname(c->readformat), ast_getformatname(c->writeformat));
- return -1;
- }
- }
+ ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n",
+ frame->subclass, c->nativeformats, c->readformat, c->writeformat);
+ return -1;
}
}
if (p) {
@@ -689,9 +580,6 @@ static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
}
ast_mutex_unlock(&p->lock);
}
- /* Free translated frame */
- if(need_frfree)
- ast_frfree(frame);
return res;
}
@@ -725,7 +613,7 @@ static int oh323_indicate(struct ast_channel *c, int condition)
}
return 0;
case -1:
- return 0;
+ return -1;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
return -1;
@@ -756,20 +644,13 @@ static struct ast_channel *oh323_new(struct oh323_pvt *i, int state, const char
if (ch) {
-// snprintf(ch->name, sizeof(ch->name)-1, "H323/%s-%04x", host, rand() & 0xffff);
snprintf(ch->name, sizeof(ch->name)-1, "H323/%s", host);
ch->nativeformats = i->capability;
if (!ch->nativeformats)
ch->nativeformats = capability;
fmt = ast_best_codec(ch->nativeformats);
ch->type = type;
-
- /* RTP stream */
ch->fds[0] = ast_rtp_fd(i->rtp);
-
- /* RTCP stream */
- ch->fds[1] = ast_rtcp_fd(i->rtp);
-
ast_setstate(ch, state);
if (state == AST_STATE_RING)
@@ -810,10 +691,6 @@ static struct ast_channel *oh323_new(struct oh323_pvt *i, int state, const char
ch->priority = 1;
if (strlen(i->callerid))
ch->callerid = strdup(i->callerid);
- if (strlen(i->dnid))
- ch->dnid = strdup(i->dnid);
- if (strlen(i->rdnis))
- ch->rdnis = strdup(i->rdnis);
if (strlen(i->accountcode))
strncpy(ch->accountcode, i->accountcode, sizeof(ch->accountcode)-1);
if (i->amaflags)
@@ -854,7 +731,6 @@ static struct oh323_pvt *oh323_alloc(int callid)
p->cd.call_reference = callid;
p->bridge = bridge_default;
- memcpy(&p->calloptions, &global_options, sizeof(global_options));
p->dtmfmode = dtmfmode;
if (p->dtmfmode & H323_DTMF_RFC2833)
@@ -872,24 +748,25 @@ static struct oh323_pvt *find_call(int call_reference)
{
struct oh323_pvt *p;
- ast_mutex_lock(&iflock);
+ ast_mutex_lock(&iflock);
p = iflist;
while(p) {
if (p->cd.call_reference == call_reference) {
/* Found the call */
- ast_mutex_unlock(&iflock);
- return p;
+ ast_mutex_unlock(&iflock);
+ return p;
}
p = p->next;
}
ast_mutex_unlock(&iflock);
-
- return NULL;
+ return NULL;
+
}
static struct ast_channel *oh323_request(char *type, int format, void *data)
{
+
int oldformat;
struct oh323_pvt *p;
struct ast_channel *tmpc = NULL;
@@ -958,45 +835,20 @@ static struct ast_channel *oh323_request(char *type, int format, void *data)
return tmpc;
}
-struct oh323_alias *find_alias(call_details_t cd)
+struct oh323_alias *find_alias(const char *source_aliases)
{
- struct oh323_alias *a, *a_e164 = NULL, *a_pfx = NULL;
- char *s, *p;
- int a_pfxlen, numlen;
+ struct oh323_alias *a;
a = aliasl.aliases;
- a_pfxlen = 0;
- numlen = strlen(cd.call_dest_e164);
while(a) {
- if (!strcasecmp(a->name, cd.call_dest_alias)) {
+ if (!strcasecmp(a->name, source_aliases)) {
break;
}
- /* Check for match of E164 number */
- if (!strcasecmp(a->e164, cd.call_dest_e164))
- a_e164 = a;
- else { /* Check for match called number with prefixes */
- for(s = a->prefix; *s; ) {
- for(; *s == ' '; ++s);
- if(!(p = strchr(s, ',')))
- p = s + strlen(s);
- if((p - s > a_pfxlen) && (numlen >= p - s) && (strncasecmp(s, cd.call_dest_e164, p - s) == 0)) {
- a_pfxlen = p - s;
- a_pfx = a;
- }
- s = p;
- if(*s == ',')
- ++s;
- }
- }
a = a->next;
}
- if(a)
- return a;
- if(a_e164)
- return a_e164;
- return a_pfx;
+ return a;
}
struct oh323_user *find_user(const call_details_t cd)
@@ -1042,27 +894,6 @@ struct oh323_peer *find_peer(char *dest_peer)
}
-static int progress(unsigned call_reference, int inband)
-{
- struct oh323_pvt *p;
-
- ast_log(LOG_DEBUG, "Received ALERT/PROGRESS message for %s tones\n", (inband ? "inband" : "self-generated"));
- p = find_call(call_reference);
-
- if (!p) {
- ast_log(LOG_ERROR, "Private structure not found in send_digit.\n");
- return -1;
- }
- if (!p->owner) {
- ast_log(LOG_ERROR, "No asterisk's channel associated with private structure.\n");
- return -1;
- }
-
- ast_queue_control(p->owner, (inband ? AST_CONTROL_PROGRESS : AST_CONTROL_RINGING), 0);
-
- return 0;
-}
-
/**
* Callback for sending digits from H.323 up to asterisk
*
@@ -1081,13 +912,13 @@ int send_digit(unsigned call_reference, char digit)
}
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
- f.subclass = digit;
- f.datalen = 0;
- f.samples = 300;
+ f.subclass = digit;
+ f.datalen = 0;
+ f.samples = 300;
f.offset = 0;
f.data = NULL;
- f.mallocd = 0;
- f.src = "SEND_DIGIT";
+ f.mallocd = 0;
+ f.src = "SEND_DIGIT";
return ast_queue_frame(p->owner, &f, 1);
}
@@ -1131,21 +962,20 @@ struct rtp_info *create_connection(unsigned call_reference)
* Returns 1 on success
*/
-call_options_t *setup_incoming_call(call_details_t cd)
+int setup_incoming_call(call_details_t cd)
{
struct oh323_pvt *p = NULL;
struct ast_channel *c = NULL;
struct oh323_user *user = NULL;
struct oh323_alias *alias = NULL;
- call_options_t *call_options = NULL;
/* allocate the call*/
p = oh323_alloc(cd.call_reference);
if (!p) {
ast_log(LOG_ERROR, "Unable to allocate private structure, this is bad.\n");
- return NULL;
+ return 0;
}
/* Populate the call details in the private structure */
@@ -1156,28 +986,21 @@ call_options_t *setup_incoming_call(call_details_t cd)
p->cd.call_dest_e164 = cd.call_dest_e164;
if (h323debug) {
- ast_verbose(VERBOSE_PREFIX_2 "Setting up Call\n");
- ast_verbose(VERBOSE_PREFIX_3 "Calling party name: [%s]\n", p->cd.call_source_aliases);
- ast_verbose(VERBOSE_PREFIX_3 "Calling party number: [%s]\n", p->cd.call_source_e164);
- ast_verbose(VERBOSE_PREFIX_3 "Called party name: [%s]\n", p->cd.call_dest_alias);
- ast_verbose(VERBOSE_PREFIX_3 "Called party number: [%s]\n", p->cd.call_dest_e164);
- ast_verbose(VERBOSE_PREFIX_3 "Redirecting party number: [%s]\n", p->cd.call_redir_e164);
+ ast_verbose(VERBOSE_PREFIX_3 "Setting up Call\n");
+ ast_verbose(VERBOSE_PREFIX_3 " Calling party name: [%s]\n", p->cd.call_source_aliases);
+ ast_verbose(VERBOSE_PREFIX_3 " Calling party number: [%s]\n", p->cd.call_source_e164);
+ ast_verbose(VERBOSE_PREFIX_3 " Called party name: [%s]\n", p->cd.call_dest_alias);
+ ast_verbose(VERBOSE_PREFIX_3 " Called party number: [%s]\n", p->cd.call_dest_e164);
}
/* Decide if we are allowing Gatekeeper routed calls*/
if ((!strcasecmp(cd.sourceIp, gatekeeper)) && (gkroute == -1) && (usingGk == 1)) {
if (strlen(cd.call_dest_e164)) {
- char *ctx;
-
- alias = find_alias(cd);
- ctx = alias ? alias->context : default_context;
-
- strncpy(p->dnid, cd.call_dest_e164, sizeof(p->dnid)-1);
- strncpy(p->exten, cd.call_dest_e164, sizeof(p->exten)-1);
- strncpy(p->context, ctx, sizeof(p->context)-1);
+ strncpy(p->exten, cd.call_dest_e164, sizeof(p->exten)-1);
+ strncpy(p->context, default_context, sizeof(p->context)-1);
} else {
- alias = find_alias(cd);
+ alias = find_alias(cd.call_dest_alias);
if (!alias) {
ast_log(LOG_ERROR, "Call for %s rejected, alias not found\n", cd.call_dest_alias);
@@ -1187,8 +1010,8 @@ call_options_t *setup_incoming_call(call_details_t cd)
strncpy(p->context, alias->context, sizeof(p->context)-1);
}
- /* Asterisk prefers user name to be quoted */
- sprintf(p->callerid, "\"%s\" <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
+
+ sprintf(p->callerid, "%s <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
} else {
/* Either this call is not from the Gatekeeper
@@ -1199,13 +1022,11 @@ call_options_t *setup_incoming_call(call_details_t cd)
if (!user) {
- /* Asterisk prefers user name to be quoted */
- sprintf(p->callerid, "\"%s\" <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
+ sprintf(p->callerid, "%s <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
if (strlen(p->cd.call_dest_e164)) {
- strncpy(p->dnid, cd.call_dest_e164, sizeof(p->dnid)-1);
strncpy(p->exten, cd.call_dest_e164, sizeof(p->exten)-1);
} else {
- strncpy(p->exten, cd.call_dest_alias, sizeof(p->exten)-1);
+ strncpy(p->exten, cd.call_dest_alias, sizeof(p->exten)-1);
}
if (!strlen(default_context)) {
ast_log(LOG_ERROR, "Call from user '%s' rejected due to no default context\n", p->cd.call_source_aliases);
@@ -1213,14 +1034,13 @@ call_options_t *setup_incoming_call(call_details_t cd)
}
strncpy(p->context, default_context, sizeof(p->context)-1);
ast_log(LOG_DEBUG, "Sending %s to context [%s]\n", cd.call_source_aliases, p->context);
- } else {
- call_options = &user->call_options;
+ } else {
if (user->host) {
if (strcasecmp(cd.sourceIp, inet_ntoa(user->addr.sin_addr))){
if(!strlen(default_context)) {
ast_log(LOG_ERROR, "Call from user '%s' rejected due to non-matching IP address of '%s'\n", user->name, cd.sourceIp);
- return NULL;
+ return 0;
}
strncpy(p->context, default_context, sizeof(p->context)-1);
@@ -1232,12 +1052,12 @@ call_options_t *setup_incoming_call(call_details_t cd)
if (user->incominglimit > 0) {
if (user->inUse >= user->incominglimit) {
ast_log(LOG_ERROR, "Call from user '%s' rejected due to usage limit of %d\n", user->name, user->incominglimit);
- return NULL;
+ return 0;
}
}
strncpy(p->context, user->context, sizeof(p->context)-1);
p->bridge = user->bridge;
- p->nat = user->nat;
+ p->nat = user->nat;
if (strlen(user->callerid))
strncpy(p->callerid, user->callerid, sizeof(p->callerid) - 1);
@@ -1260,21 +1080,14 @@ call_options_t *setup_incoming_call(call_details_t cd)
/* I know this is horrid, don't kill me saddam */
exit:
-// if(strlen(p->cd.call_redir_e164))
-// strncpy(p->rdnis, cd.call_redir_e164, sizeof(p->rdnis)-1);
/* allocate a channel and tell asterisk about it */
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 NULL;
+ return 0;
}
- ast_queue_control(c, AST_CONTROL_RINGING, 0);
-
- if(!call_options)
- return &global_options;
-
- return call_options;
+ return 1;
}
/**
@@ -1304,12 +1117,10 @@ if (!p) {
*
* Returns nothing
*/
-void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort, int direction, int payloadType)
+void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort)
{
struct oh323_pvt *p = NULL;
struct sockaddr_in them;
- struct rtpPayloadType payload;
- struct ast_channel *chan;
/* Find the call or allocate a private structure if call not found */
p = find_call(call_reference);
@@ -1322,56 +1133,11 @@ void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int rem
them.sin_family = AF_INET;
them.sin_addr.s_addr = inet_addr(remoteIp); // only works for IPv4
them.sin_port = htons(remotePort);
-
- /* Find RTP payload <=> asterisk's subcode association */
- payload = ast_rtp_lookup_pt(p->rtp, payloadType);
-
- ast_log(LOG_DEBUG, "Setting up %sbound RTP connection for %s:%d with payload %s (RTP code %d)\n", (direction ? "out" : "in"), inet_ntoa(them.sin_addr), remotePort, ast_getformatname(payload.code), payloadType);
-
- /* Set channel's native codec and prepare translation table
- * for given direction and currently used format
- */
- if ((chan = p->owner)) {
- if (payload.isAstFormat) {
- /* Don't allow any transmission until codec is changed */
-// ast_mutex_lock(&chan->lock);
- chan->nativeformats = payload.code;
- if(direction)
- ast_set_write_format(chan, chan->writeformat);
- else
- ast_set_read_format(chan, chan->readformat);
-// ast_mutex_unlock(&chan->lock);
- }
- }
-
ast_rtp_set_peer(p->rtp, &them);
-
- if(p->calloptions.progress_audio)
- progress(call_reference, TRUE);
return;
}
-/* Not used for now - set RTP peer's address */
-void setup_rtp_peer(unsigned call_reference, const char *remoteIp, int remotePort)
-{
- struct oh323_pvt *p = NULL;
- struct sockaddr_in them;
-
- /* Find the call or allocate a private structure if call not found */
- p = find_call(call_reference);
-
- if (!p) {
- ast_log(LOG_ERROR, "Something is wrong: rtp\n");
- return;
- }
-
- them.sin_family = AF_INET;
- them.sin_addr.s_addr = inet_addr(remoteIp); // only works for IPv4
- them.sin_port = htons(remotePort);
- ast_rtp_set_peer(p->rtp, &them);
-}
-
/**
* Call-back function to signal asterisk that the channel has been answered
* Returns nothing
@@ -1470,15 +1236,13 @@ restartsearch:
}
ast_mutex_unlock(&iflock);
+ pthread_testcancel();
+
/* Wait for sched or io */
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
res = 1000;
res = ast_io_wait(io, res);
-
- /* Check for thread cancellation */
- pthread_testcancel();
-
ast_mutex_lock(&monlock);
if (res >= 0)
ast_sched_runq(sched);
@@ -1628,7 +1392,7 @@ int reload_config(void)
struct oh323_alias *alias = NULL;
struct hostent *hp;
char *cat;
- char *utype;
+ char *utype;
cfg = ast_load(config);
@@ -1644,8 +1408,6 @@ int reload_config(void)
}
h323debug=0;
dtmfmode = H323_DTMF_RFC2833;
- /* Fill global variables with pre-determined values */
- memset(&global_options, 0, sizeof(global_options));
memset(&bindaddr, 0, sizeof(bindaddr));
@@ -1716,33 +1478,6 @@ int reload_config(void)
ast_log(LOG_WARNING, "Unknown dtmf mode '%s', using rfc2833\n", v->value);
dtmfmode = H323_DTMF_RFC2833;
}
- /* Setup global parameters */
- } else if (!strcasecmp(v->name, "noFastStart")) {
- global_options.noFastStart = ast_true(v->value);
- } else if (!strcasecmp(v->name, "noH245Tunneling")) {
- global_options.noH245Tunnelling = ast_true(v->value);
- } else if (!strcasecmp(v->name, "noSilenceSuppression")) {
- global_options.noSilenceSuppression = ast_true(v->value);
- } else if (!strcasecmp(v->name, "progress_setup")) {
- int progress_setup = atoi(v->value);
- if((progress_setup != 0) &&
- (progress_setup != 1) &&
- (progress_setup != 3) &&
- (progress_setup != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_setup at line %d, assuming 0\n", progress_setup, v->lineno);
- progress_setup = 0;
- }
- global_options.progress_setup = progress_setup;
- } else if (!strcasecmp(v->name, "progress_alert")) {
- int progress_alert = atoi(v->value);
- if((progress_alert != 0) &&
- (progress_alert != 8)) {
- ast_log(LOG_WARNING, "Invalid value %d for progress_alert at line %d, assuming 0\n", progress_alert, v->lineno);
- progress_alert = 0;
- }
- global_options.progress_alert = progress_alert;
- } else if (!strcasecmp(v->name, "progress_audio")) {
- global_options.progress_audio = ast_true(v->value);
} else if (!strcasecmp(v->name, "UserByAlias")) {
userbyalias = ast_true(v->value);
} else if (!strcasecmp(v->name, "bridge")) {
@@ -1873,20 +1608,15 @@ int reload(void)
delete_aliases();
prune_peers();
- reload_config();
-
-
-#if 0
-
-This code causes seg on -r
-
if (strlen(gatekeeper)) {
h323_gk_urq();
}
+ reload_config();
+#if 0
/* Possibly register with a GK */
- if (!gatekeeper_disable) {
+ if (gatekeeper_disable == 0) {
if (h323_set_gk(gatekeeper_discover, gatekeeper, secret)) {
ast_log(LOG_ERROR, "Gatekeeper registration failed.\n");
h323_end_process();
@@ -1894,7 +1624,6 @@ This code causes seg on -r
}
}
#endif
-
restart_monitor();
return 0;
}
@@ -2018,9 +1747,7 @@ int load_module()
create_connection,
setup_rtp_connection,
cleanup_connection,
- connection_made,
- send_digit,
- progress);
+ connection_made, send_digit);
/* start the h.323 listener */
@@ -2078,7 +1805,6 @@ int unload_module()
return -1;
}
h323_gk_urq();
- h323_debug(0,0);
h323_end_process();
/* unregister rtp */
@@ -2117,3 +1843,7 @@ char *key()
{
return ASTERISK_GPL_KEY;
}
+
+
+
+