[Asterisk-cvs] asterisk/channels chan_h323.c,1.16,1.17
jeremy at lists.digium.com
jeremy at lists.digium.com
Tue Dec 23 17:09:49 CST 2003
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Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv4599
Modified Files:
chan_h323.c
Log Message:
Apply massive patch from PCadach. If things are broken blame him. Bug#469
Index: chan_h323.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_h323.c,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- chan_h323.c 19 Dec 2003 18:49:17 -0000 1.16
+++ chan_h323.c 23 Dec 2003 23:01:24 -0000 1.17
@@ -46,6 +46,7 @@
#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>
@@ -59,6 +60,19 @@
#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";
@@ -106,6 +120,8 @@
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 */
@@ -245,17 +261,37 @@
} else if (!strcasecmp(v->name, "nat")) {
user->nat = ast_true(v->value);
} else if (!strcasecmp(v->name, "noFastStart")) {
- user->noFastStart = ast_true(v->value);
+ user->call_options.noFastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "noH245Tunneling")) {
- user->noH245Tunneling = ast_true(v->value);
+ user->call_options.noH245Tunnelling = ast_true(v->value);
} else if (!strcasecmp(v->name, "noSilenceSuppression")) {
- user->noSilenceSuppression = ast_true(v->value);
+ user->call_options.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)
@@ -332,11 +368,31 @@
} else if (!strcasecmp(v->name, "bridge")) {
peer->bridge = ast_true(v->value);
} else if (!strcasecmp(v->name, "noFastStart")) {
- peer->noFastStart = ast_true(v->value);
+ peer->call_options.noFastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "noH245Tunneling")) {
- peer->noH245Tunneling = ast_true(v->value);
+ peer->call_options.noH245Tunnelling = ast_true(v->value);
} else if (!strcasecmp(v->name, "noSilenceSuppression")) {
- peer->noSilenceSuppression = ast_true(v->value);
+ 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);
} else if (!strcasecmp(v->name, "outgoinglimit")) {
peer->outgoinglimit = atoi(v->value);
if (peer->outgoinglimit > 0)
@@ -427,16 +483,16 @@
} 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_RINGING);
+ ast_setstate(c, AST_STATE_RING);
return 0;
}
@@ -504,20 +560,31 @@
return 0;
}
-static struct ast_frame *oh323_rtp_read(struct oh323_pvt *p)
+/* Pass channel struct too to allow RTCP handling */
+static struct ast_frame *oh323_rtp_read(struct ast_channel *c, 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;
- }
+ /* 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;
+ }
- 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;
@@ -525,10 +592,12 @@
/* 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);
- ast_set_write_format(p->owner, p->owner->writeformat);
+ /* Don't set write format because it will be set up when channel started */
+// ast_set_write_format(p->owner, p->owner->writeformat);
}
/* Do in-band DTMF detection */
@@ -550,7 +619,8 @@
struct ast_frame *fr;
struct oh323_pvt *p = c->pvt->pvt;
ast_mutex_lock(&p->lock);
- fr = oh323_rtp_read(p);
+ /* Pass channel structure to handle other streams than just RTP */
+ fr = oh323_rtp_read(c, p);
ast_mutex_unlock(&p->lock);
return fr;
}
@@ -559,6 +629,7 @@
{
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;
@@ -568,9 +639,47 @@
}
} else {
if (!(frame->subclass & c->nativeformats)) {
- 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(!(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;
+ }
+ }
}
}
if (p) {
@@ -580,6 +689,9 @@
}
ast_mutex_unlock(&p->lock);
}
+ /* Free translated frame */
+ if(need_frfree)
+ ast_frfree(frame);
return res;
}
@@ -613,7 +725,7 @@
}
return 0;
case -1:
- return -1;
+ return 0;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
return -1;
@@ -644,13 +756,20 @@
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)
@@ -691,6 +810,10 @@
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)
@@ -731,6 +854,7 @@
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)
@@ -748,25 +872,24 @@
{
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;
@@ -835,20 +958,45 @@
return tmpc;
}
-struct oh323_alias *find_alias(const char *source_aliases)
+struct oh323_alias *find_alias(call_details_t cd)
{
- struct oh323_alias *a;
+ struct oh323_alias *a, *a_e164 = NULL, *a_pfx = NULL;
+ char *s, *p;
+ int a_pfxlen, numlen;
a = aliasl.aliases;
+ a_pfxlen = 0;
+ numlen = strlen(cd.call_dest_e164);
while(a) {
- if (!strcasecmp(a->name, source_aliases)) {
+ if (!strcasecmp(a->name, cd.call_dest_alias)) {
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;
}
- return a;
+ if(a)
+ return a;
+ if(a_e164)
+ return a_e164;
+ return a_pfx;
}
struct oh323_user *find_user(const call_details_t cd)
@@ -894,6 +1042,27 @@
}
+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
*
@@ -912,13 +1081,13 @@
}
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);
}
@@ -962,20 +1131,21 @@
* Returns 1 on success
*/
-int setup_incoming_call(call_details_t cd)
+call_options_t *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 0;
+ return NULL;
}
/* Populate the call details in the private structure */
@@ -986,21 +1156,28 @@
p->cd.call_dest_e164 = cd.call_dest_e164;
if (h323debug) {
- 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);
+ 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);
}
/* Decide if we are allowing Gatekeeper routed calls*/
if ((!strcasecmp(cd.sourceIp, gatekeeper)) && (gkroute == -1) && (usingGk == 1)) {
if (strlen(cd.call_dest_e164)) {
- strncpy(p->exten, cd.call_dest_e164, sizeof(p->exten)-1);
- strncpy(p->context, default_context, sizeof(p->context)-1);
+ 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);
} else {
- alias = find_alias(cd.call_dest_alias);
+ alias = find_alias(cd);
if (!alias) {
ast_log(LOG_ERROR, "Call for %s rejected, alias not found\n", cd.call_dest_alias);
@@ -1010,8 +1187,8 @@
strncpy(p->context, alias->context, sizeof(p->context)-1);
}
-
- sprintf(p->callerid, "%s <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
+ /* Asterisk prefers user name to be quoted */
+ sprintf(p->callerid, "\"%s\" <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
} else {
/* Either this call is not from the Gatekeeper
@@ -1022,11 +1199,13 @@
if (!user) {
- sprintf(p->callerid, "%s <%s>", p->cd.call_source_aliases, p->cd.call_source_e164);
+ /* Asterisk prefers user name to be quoted */
+ 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);
@@ -1034,13 +1213,14 @@
}
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 {
+ } else {
+ call_options = &user->call_options;
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 0;
+ return NULL;
}
strncpy(p->context, default_context, sizeof(p->context)-1);
@@ -1052,12 +1232,12 @@
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 0;
+ return NULL;
}
}
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);
@@ -1080,14 +1260,21 @@
/* 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 0;
+ return NULL;
}
- return 1;
+ ast_queue_control(c, AST_CONTROL_RINGING, 0);
+
+ if(!call_options)
+ return &global_options;
+
+ return call_options;
}
/**
@@ -1117,10 +1304,12 @@
*
* Returns nothing
*/
-void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort)
+void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort, int direction, int payloadType)
{
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);
@@ -1133,11 +1322,56 @@
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
@@ -1236,13 +1470,15 @@
}
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);
@@ -1392,7 +1628,7 @@
struct oh323_alias *alias = NULL;
struct hostent *hp;
char *cat;
- char *utype;
+ char *utype;
cfg = ast_load(config);
@@ -1408,6 +1644,8 @@
}
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));
@@ -1478,6 +1716,33 @@
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")) {
@@ -1608,15 +1873,20 @@
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 == 0) {
+ if (!gatekeeper_disable) {
if (h323_set_gk(gatekeeper_discover, gatekeeper, secret)) {
ast_log(LOG_ERROR, "Gatekeeper registration failed.\n");
h323_end_process();
@@ -1624,6 +1894,7 @@
}
}
#endif
+
restart_monitor();
return 0;
}
@@ -1747,7 +2018,9 @@
create_connection,
setup_rtp_connection,
cleanup_connection,
- connection_made, send_digit);
+ connection_made,
+ send_digit,
+ progress);
/* start the h.323 listener */
@@ -1805,6 +2078,7 @@
return -1;
}
h323_gk_urq();
+ h323_debug(0,0);
h323_end_process();
/* unregister rtp */
@@ -1843,7 +2117,3 @@
{
return ASTERISK_GPL_KEY;
}
-
-
-
-
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