Remove unnecessary (long time ago commented out) code

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index c3368aded990e8338bdf8efa8165cf4486630c9e..529a00b513acfd5f08d0e2693a71fb0a36339893 100644 (file)
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -1081,10 +1081,6 @@ static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const c
                        ch->cid.cid_dnid = strdup(pvt->exten);
                }
                ast_setstate(ch, state);
-#if 0
-               if (pvt->rtp)
-                       ast_jb_configure(ch, &global_jbconf);
-#endif
                if (state != AST_STATE_DOWN) {
                        if (ast_pbx_start(ch)) {
                                ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ch->name);
@@ -1109,15 +1105,6 @@ static struct oh323_pvt *oh323_alloc(int callid)
        }
        memset(pvt, 0, sizeof(struct oh323_pvt));
        pvt->cd.redirect_reason = -1;
-#if 0
-       pvt->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0,bindaddr.sin_addr);
-       if (!pvt->rtp) {
-               ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
-               free(pvt);
-               return NULL;
-       }
-       ast_rtp_settos(pvt->rtp, tos);
-#endif
        /* Ensure the call token is allocated for outgoing call */
        if (!callid) {
                if ((pvt->cd).call_token == NULL) {
@@ -1625,13 +1612,6 @@ static int create_addr(struct oh323_pvt *pvt, char *opeer)
                found++;
                memcpy(&pvt->options, &p->options, sizeof(pvt->options));
                pvt->jointcapability = pvt->options.capability;
-#if 0
-               if (pvt->rtp) {
-                       if (h323debug)
-                               ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", pvt->options.nat);
-                       ast_rtp_setnat(pvt->rtp, pvt->options.nat);
-               }
-#endif
                if (pvt->options.dtmfmode) {
                        if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
                                pvt->nonCodecCapability |= AST_RTP_DTMF;
@@ -1663,13 +1643,6 @@ static int create_addr(struct oh323_pvt *pvt, char *opeer)
                        if (p) {
                                ASTOBJ_UNREF(p, oh323_destroy_peer);
                        }
-#if 0
-                       if (pvt->rtp) {
-                               if (h323debug)
-                                       ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", pvt->options.nat);
-                               ast_rtp_setnat(pvt->rtp, pvt->options.nat);
-                       }
-#endif
                        if (pvt->options.dtmfmode) {
                                if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
                                        pvt->nonCodecCapability |= AST_RTP_DTMF;
@@ -1748,13 +1721,6 @@ static struct ast_channel *oh323_request(const char *type, int format, void *dat
        else {
                memcpy(&pvt->options, &global_options, sizeof(pvt->options));
                pvt->jointcapability = pvt->options.capability;
-#if 0
-               if (pvt->rtp) {
-                       if (h323debug)
-                               ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", pvt->options.nat);
-                       ast_rtp_setnat(pvt->rtp, pvt->options.nat);
-               }
-#endif
                if (pvt->options.dtmfmode) {
                        if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
                                pvt->nonCodecCapability |= AST_RTP_DTMF;

diff --git a/channels/h323/ast_h323.h b/channels/h323/ast_h323.h
index 946811be84dda125269fbeac7b86208e88729c7e..2d499ac3af875a67fc8a51c89dc6b781cbed51fc 100644 (file)
--- a/channels/h323/ast_h323.h
+++ b/channels/h323/ast_h323.h
@@ -31,33 +31,6 @@
 
 #define VERSION(a,b,c) ((a)*10000+(b)*100+(c))
 
-#if 0
-/**  These need to be redefined here because the C++
-     side of this driver is blind to the asterisk headers */
-/*! G.723.1 compression */
-#define AST_FORMAT_G723_1      (1 << 0)
-/*! GSM compression */
-#define AST_FORMAT_GSM         (1 << 1)
-/*! Raw mu-law data (G.711) */
-#define AST_FORMAT_ULAW                (1 << 2)
-/*! Raw A-law data (G.711) */
-#define AST_FORMAT_ALAW                (1 << 3)
-/*! MPEG-2 layer 3 */
-#define AST_FORMAT_MP3         (1 << 4)
-/*! ADPCM (whose?) */
-#define AST_FORMAT_ADPCM       (1 << 5)
-/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
-#define AST_FORMAT_SLINEAR     (1 << 6)
-/*! LPC10, 180 samples/frame */
-#define AST_FORMAT_LPC10       (1 << 7)
-/*! G.729A audio */
-#define AST_FORMAT_G729A       (1 << 8)
-/*! SpeeX Free Compression */
-#define AST_FORMAT_SPEEX       (1 << 9)
-/*! ILBC Free Codec */
-#define AST_FORMAT_ILBC                (1 << 10)
-#endif
-
 /**This class describes the G.711 codec capability.
  */
 class AST_G711Capability : public H323AudioCapability