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Version 0.1.5 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2000-12-14 13:50:41 +00:00
parent fa5bb58d49
commit ce46416cb3
3 changed files with 24 additions and 13 deletions
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3
CHANGES
View File

@@ -1,3 +1,6 @@
* Asterisk 0.1.5
-- Fix a small mistake in IAX
-- Fix the QuickNet driver to work with newer cards
* Asterisk 0.1.4
-- Update VoFR some more
-- Fix the QuickNet driver to work with LineJack

2
channels/chan_iax.c
View File

@@ -1893,7 +1893,7 @@ static struct ast_channel *iax_request(char *type, int format, void *data)
char *st;
struct ast_channel *c;
strncpy(s, (char *)data, sizeof(s));
strtok(s, ":");
strtok(s, "/");
strtok(s, "@");
st = strtok(NULL, "@");
if (!st)

32
channels/chan_phone.c
View File

@@ -284,7 +284,8 @@ static struct ast_frame *phone_read(struct ast_channel *ast)
phonee.bytes = ioctl(p->fd, PHONE_EXCEPTION);
if (phonee.bits.dtmf_ready) {
ast_log(LOG_DEBUG, "phone_read(): DTMF\n");
if (option_debug)
ast_log(LOG_DEBUG, "phone_read(): DTMF\n");
/* We've got a digit -- Just handle this nicely and easily */
digit = ioctl(p->fd, PHONE_GET_DTMF_ASCII);
@@ -293,10 +294,12 @@ static struct ast_frame *phone_read(struct ast_channel *ast)
return &p->fr;
}
if (phonee.bits.hookstate) {
ast_log(LOG_DEBUG, "Hookstate changed\n");
if (option_debug)
ast_log(LOG_DEBUG, "Hookstate changed\n");
res = ioctl(p->fd, PHONE_HOOKSTATE);
/* See if we've gone on hook, if so, notify by returning NULL */
ast_log(LOG_DEBUG, "New hookstate: %d\n", res);
if (option_debug)
ast_log(LOG_DEBUG, "New hookstate: %d\n", res);
if (!res && (p->mode != MODE_FXO))
return NULL;
else {
@@ -396,6 +399,7 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
char *pos;
int sofar;
int expected;
int codecset = 0;
char tmpbuf[4];
/* Write a frame of (presumably voice) data */
if (frame->frametype != AST_FRAME_VOICE) {
@@ -429,6 +433,7 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
p->lastinput = AST_FORMAT_G723_1;
/* Reset output buffer */
p->obuflen = 0;
codecset = 1;
}
if (frame->datalen > 24) {
ast_log(LOG_WARNING, "Frame size too large for G.723.1 (%d bytes)\n", frame->datalen);
@@ -449,20 +454,23 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
}
p->lastformat = AST_FORMAT_SLINEAR;
p->lastinput = AST_FORMAT_SLINEAR;
codecset = 1;
/* Reset output buffer */
p->obuflen = 0;
}
maxfr = 480;
}
ioctl(p->fd, PHONE_REC_DEPTH, 3);
ioctl(p->fd, PHONE_PLAY_DEPTH, 3);
if (ioctl(p->fd, PHONE_PLAY_START)) {
ast_log(LOG_WARNING, "Failed to start playback\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_START)) {
ast_log(LOG_WARNING, "Failed to start recording\n");
return -1;
if (codecset) {
ioctl(p->fd, PHONE_REC_DEPTH, 3);
ioctl(p->fd, PHONE_PLAY_DEPTH, 3);
if (ioctl(p->fd, PHONE_PLAY_START)) {
ast_log(LOG_WARNING, "Failed to start playback\n");
return -1;
}
if (ioctl(p->fd, PHONE_REC_START)) {
ast_log(LOG_WARNING, "Failed to start recording\n");
return -1;
}
}
/* If we get here, we have a voice frame of Appropriate data */
sofar = 0;