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-rwxr-xr-xchannels/chan_agent.c2
-rwxr-xr-xchannels/chan_alsa.c14
-rwxr-xr-xchannels/chan_iax2.c12
-rwxr-xr-xchannels/chan_mgcp.c22
-rwxr-xr-xchannels/chan_modem_i4l.c16
-rwxr-xr-xchannels/chan_skinny.c6
6 files changed, 36 insertions, 36 deletions
diff --git a/channels/chan_agent.c b/channels/chan_agent.c
index c953ba389..402c0949c 100755
--- a/channels/chan_agent.c
+++ b/channels/chan_agent.c
@@ -508,7 +508,7 @@ static int agent_call(struct ast_channel *ast, char *dest, int timeout)
if (res)
ast_log(LOG_WARNING, "Unable to set read format to %s\n", ast_getformatname(ast_best_codec(p->chan->nativeformats)));
} else {
- // Agent hung-up
+ /* Agent hung-up */
p->chan = NULL;
}
diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index d0eb84478..bfb566d4e 100755
--- a/channels/chan_alsa.c
+++ b/channels/chan_alsa.c
@@ -1,9 +1,9 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
- * Copyright (C) 2002, Linux Support Services
+ * Copyright (C) 2002-2004, Digium, inc
*
- * By Matthew Fredrickson <creslin@linux-support.net>
+ * By Matthew Fredrickson <creslin@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
@@ -61,7 +61,7 @@
#define MIN_SWITCH_TIME 600
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
-//static int block = O_NONBLOCK;
+/* static int block = O_NONBLOCK; */
static char indevname[50] = ALSA_INDEV;
static char outdevname[50] = ALSA_OUTDEV;
@@ -284,14 +284,14 @@ static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
snd_pcm_sw_params_t *swparams = NULL;
struct pollfd pfd;
snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
- //int period_bytes = 0;
+ /* int period_bytes = 0; */
snd_pcm_uframes_t buffer_size = 0;
unsigned int rate = DESIRED_RATE;
#if 0
unsigned int per_min = 1;
#endif
- //unsigned int per_max = 8;
+ /* unsigned int per_max = 8; */
snd_pcm_uframes_t start_threshold, stop_threshold;
err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK);
@@ -334,7 +334,7 @@ static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
ast_log(LOG_DEBUG, "Period size is %d\n", err);
}
- buffer_size = 4096 * 2; //period_size * 16;
+ buffer_size = 4096 * 2; /* period_size * 16; */
err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
if (err < 0) {
ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
@@ -542,7 +542,7 @@ static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
int len = sizpos;
int pos;
int res = 0;
- //size_t frames = 0;
+ /* size_t frames = 0; */
snd_pcm_state_t state;
/* Immediately return if no sound is enabled */
if (nosound)
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index edc7293be..7eb4d1bdc 100755
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -6617,8 +6617,8 @@ static struct iax2_peer *build_peer(const char *name, struct ast_variable *v, in
}
} else if (!strcasecmp(v->name, "timezone")) {
strncpy(peer->zonetag, v->value, sizeof(peer->zonetag)-1);
- }// else if (strcasecmp(v->name,"type"))
- // ast_log(LOG_WARNING, "Ignoring %s\n", v->name);
+ }/* else if (strcasecmp(v->name,"type")) */
+ /* ast_log(LOG_WARNING, "Ignoring %s\n", v->name); */
v=v->next;
}
if (!peer->authmethods)
@@ -6755,8 +6755,8 @@ static struct iax2_user *build_user(const char *name, struct ast_variable *v, in
}
} else if (!strcasecmp(v->name, "inkeys")) {
strncpy(user->inkeys, v->value, sizeof(user->inkeys) - 1);
- }// else if (strcasecmp(v->name,"type"))
- // ast_log(LOG_WARNING, "Ignoring %s\n", v->name);
+ }/* else if (strcasecmp(v->name,"type")) */
+ /* ast_log(LOG_WARNING, "Ignoring %s\n", v->name); */
v = v->next;
}
if (!user->authmethods) {
@@ -7038,8 +7038,8 @@ static int set_config(char *config_file, struct sockaddr_in* sin){
}
} else if (!strcasecmp(v->name, "language")) {
strncpy(language, v->value, sizeof(language) - 1);
- } //else if (strcasecmp(v->name,"type"))
- // ast_log(LOG_WARNING, "Ignoring %s\n", v->name);
+ } /*else if (strcasecmp(v->name,"type")) */
+ /* ast_log(LOG_WARNING, "Ignoring %s\n", v->name); */
v = v->next;
}
iax2_capability = capability;
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 438aefacd..e93eb04ba 100755
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1019,7 +1019,7 @@ static int mgcp_show_endpoints(int fd, int argc, char *argv[])
e = g->endpoints;
ast_cli(fd, "Gateway '%s' at %s (%s)\n", g->name, g->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), g->addr.sin_addr) : ast_inet_ntoa(iabuf, sizeof(iabuf), g->defaddr.sin_addr), g->dynamic ? "Dynamic" : "Static");
while(e) {
- // JS: Don't show wilcard endpoint
+ /* JS: Don't show wilcard endpoint */
if (strcmp(e->name, g->wcardep) !=0)
ast_cli(fd, " -- '%s@%s in '%s' is %s\n", e->name, g->name, e->context, e->sub->owner ? "active" : "idle");
hasendpoints = 1;
@@ -1721,7 +1721,7 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
#if 0
printf("Peer RTP is at port %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
#endif
- // Scan through the RTP payload types specified in a "m=" line:
+ /* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(sub->rtp);
codecs = m + len;
while(strlen(codecs)) {
@@ -1733,18 +1733,18 @@ static int process_sdp(struct mgcp_subchannel *sub, struct mgcp_request *req)
codecs += len;
}
- // Next, scan through each "a=rtpmap:" line, noting each
- // specified RTP payload type (with corresponding MIME subtype):
+ /* Next, scan through each "a=rtpmap:" line, noting each */
+ /* specified RTP payload type (with corresponding MIME subtype): */
sdpLineNum_iterator_init(&iterator);
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
- char* mimeSubtype = strdup(a); // ensures we have enough space
+ char* mimeSubtype = strdup(a); /* ensures we have enough space */
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
- // Note: should really look at the 'freq' and '#chans' params too
+ /* Note: should really look at the 'freq' and '#chans' params too */
ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype);
free(mimeSubtype);
}
- // Now gather all of the codecs that were asked for:
+ /* Now gather all of the codecs that were asked for: */
ast_rtp_get_current_formats(sub->rtp,
&peercapability, &peerNonCodecCapability);
p->capability = capability & peercapability;
@@ -2167,7 +2167,7 @@ static int transmit_audit_endpoint(struct mgcp_endpoint *p)
struct mgcp_request resp;
reqprep(&resp, p, "AUEP");
/* SC: removed unknown param VS */
- //add_header(&resp, "F", "A,R,D,S,X,N,I,T,O,ES,E,MD,M");
+ /*add_header(&resp, "F", "A,R,D,S,X,N,I,T,O,ES,E,MD,M"); */
add_header(&resp, "F", "A");
/* SC: fill in new fields */
resp.cmd = MGCP_CMD_AUEP;
@@ -2923,7 +2923,7 @@ static int handle_request(struct mgcp_subchannel *sub, struct mgcp_request *req,
if (option_verbose > 2 && (strcmp(p->name, p->parent->wcardep) != 0)) {
ast_verbose(VERBOSE_PREFIX_3 "Resetting interface %s@%s\n", p->name, p->parent->name);
}
- // JS: For RSIP on wildcard we reset all endpoints
+ /* JS: For RSIP on wildcard we reset all endpoints */
if (!strcmp(p->name, p->parent->wcardep)) {
/* Reset all endpoints */
struct mgcp_endpoint *tmp_ep;
@@ -2931,7 +2931,7 @@ static int handle_request(struct mgcp_subchannel *sub, struct mgcp_request *req,
g = p->parent;
tmp_ep = g->endpoints;
while (tmp_ep) {
- //if ((strcmp(tmp_ep->name, "*") != 0) && (strcmp(tmp_ep->name, "aaln/*") != 0)) {
+ /*if ((strcmp(tmp_ep->name, "*") != 0) && (strcmp(tmp_ep->name, "aaln/*") != 0)) { */
if (strcmp(tmp_ep->name, g->wcardep) != 0) {
struct mgcp_subchannel *tmp_sub, *first_sub;
if (option_verbose > 2) {
@@ -3635,7 +3635,7 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
e->needaudit = 1;
}
strncpy(gw->wcardep, v->value, sizeof(gw->wcardep)-1);
- //strncpy(e->name, "aaln/*", sizeof(e->name) - 1);
+ /*strncpy(e->name, "aaln/*", sizeof(e->name) - 1); */
/* XXX Should we really check for uniqueness?? XXX */
strncpy(e->context, context, sizeof(e->context) - 1);
strncpy(e->cid_num, cid_num, sizeof(e->cid_num) - 1);
diff --git a/channels/chan_modem_i4l.c b/channels/chan_modem_i4l.c
index 48e76aa69..0fbf5d49c 100755
--- a/channels/chan_modem_i4l.c
+++ b/channels/chan_modem_i4l.c
@@ -172,8 +172,8 @@ static int i4l_init(struct ast_modem_pvt *p)
if (strlen(p->incomingmsn)) {
char *q;
snprintf(cmd, sizeof(cmd), "AT&L%s", p->incomingmsn);
- // translate , into ; since that is the seperator I4L uses, but can't be directly
- // put in the config file because it will interpret the rest of the line as comment.
+ /* translate , into ; since that is the seperator I4L uses, but can't be directly */
+ /* put in the config file because it will interpret the rest of the line as comment. */
q = cmd+4;
while (*q) {
if (*q == ',') *q = ';';
@@ -319,9 +319,9 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
/* Read the first two bytes, first, in case it's a control message */
res = read(p->fd, result, 2);
if (res < 2) {
- // short read, means there was a hangup?
- // (or is this also possible without hangup?)
- // Anyway, reading from unitialized buffers is a bad idea anytime.
+ /* short read, means there was a hangup? */
+ /* (or is this also possible without hangup?) */
+ /* Anyway, reading from unitialized buffers is a bad idea anytime. */
if (errno == EAGAIN)
return i4l_handle_escape(p, 0);
return NULL;
@@ -337,7 +337,7 @@ static struct ast_frame *i4l_read(struct ast_modem_pvt *p)
ast_modem_trim(result);
if (!strcasecmp(result, "VCON")) {
/* If we're in immediate mode, reply now */
-// if (p->mode == MODEM_MODE_IMMEDIATE)
+/* if (p->mode == MODEM_MODE_IMMEDIATE) */
return i4l_handle_escape(p, 'X');
} else
if (!strcasecmp(result, "BUSY")) {
@@ -587,11 +587,11 @@ static int i4l_dial(struct ast_modem_pvt *p, char *stuff)
char tmpmsn[255];
struct ast_channel *c = p->owner;
- // Find callerid number first, to set the correct A number
+ /* Find callerid number first, to set the correct A number */
if (c && c->cid.cid_num && !(c->cid.cid_pres & 0x20)) {
snprintf(tmpmsn, sizeof(tmpmsn), ",%s,", c->cid.cid_num);
if(strlen(p->outgoingmsn) && strstr(p->outgoingmsn,tmpmsn) != NULL) {
- // Tell ISDN4Linux to use this as A number
+ /* Tell ISDN4Linux to use this as A number */
snprintf(cmd, sizeof(cmd), "AT&E%s\n", c->cid.cid_num);
if (ast_modem_send(p, cmd, strlen(cmd))) {
ast_log(LOG_WARNING, "Unable to set A number to %s\n", c->cid.cid_num);
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index dbf27977b..9a3fa486a 100755
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3,7 +3,7 @@
*
* Implementation of the Skinny protocol
*
- * Asterisk is Copyright (C) 1999-2003 Mark Spencer
+ * Copyright (C) 1999-2004 Digium, inc
*
* chan_skinny was developed by Jeremy McNamara & Florian Overkamp
*
@@ -1449,7 +1449,7 @@ static void *skinny_ss(void *data)
((exten[0] != '*') || (!ast_strlen_zero(exten) > 2))) {
ast_log(LOG_WARNING, "Can't match [%s] from '%s' in context %s\n", exten, chan->cid.cid_num ? chan->cid.cid_num : "<Unknown Caller>", chan->context);
transmit_tone(s, SKINNY_REORDER);
- sleep(3); // hang out for 3 seconds to let congestion play
+ sleep(3); /* hang out for 3 seconds to let congestion play */
break;
}
if (!timeout)
@@ -2716,7 +2716,7 @@ void delete_devices(void)
int reload(void)
{
#if 0
-// XXX Causes Seg
+/* XXX Causes Seg - needs to be fixed, or? */
delete_devices();
reload_config();