From aa92ebffeafa55788dcebaac9b561a5248ae8810 Mon Sep 17 00:00:00 2001
From: Paul Cadach <paul@odt.east.telecom.kz>
Date: Wed, 20 Sep 2006 18:08:42 +0000
Subject: [PATCH] Remove unnecessary (long time ago commented out) code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_h323.c | 34 ----------------------------------
channels/h323/ast_h323.h | 27 ---------------------------
2 files changed, 61 deletions(-)
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index c3368aded9..529a00b513 100644
--- 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 946811be84..2d499ac3af 100644
--- 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
--
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