Displaying 9 results from an estimated 9 matches for "ast_frame_voice".
2015 Jul 07
2
Bug in ast_frame_adjust_volume in 12.2.0?
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
0 <repeats 64 times>}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64,
src = 0x51623b0 "func_jitterbuffe...
2017 Apr 06
2
Issues with Siren14 codec in Asterisk 14.3.0
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 232, offset = 64,
src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378,
uint32 = 1017877368, pad = "x\223\253<\000\000\000"},...
2004 Oct 05
1
Brazillian Caller ID: almost there...
...y ztmonitor. wcfxo correctly recognize the "DTMF
CLIP" and asterisk shot the AST_STATE_PRERING correctly.
But the DTMF tones are not reconized. In the chan_zap.c, the code:
if (f->frametype == AST_FRAME_DTMF) {
(...)
Does not occurs because the frametype is always reconized as voice
(AST_FRAME_VOICE).
I use 4 Digium X100P.
Like noted by Soren (http://www.ad2.com.br/DTMF.jpg), the main diference
between the 2 samples is the time elapsed after the "burst" sign and the
first DTMF digit. In my sample, its occurs more quickly...
I inspect the asterisk code for days, but nothing found......
2017 Apr 12
2
More issues with Siren14 datalen == 0 packets
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
uint32 = 2156475160, pad = "\030\063\211\200\000\000\0...
2005 Jan 31
1
chan_sccp bug / problem
...1
Reading symbols from /usr/lib/asterisk/modules/chan_sccp.so...done.
Loaded symbols for /usr/lib/asterisk/modules/chan_sccp.so
Reading symbols from /lib64/libgcc_s.so.1...done.
Loaded symbols for /lib64/libgcc_s.so.1
#0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
38 if (f->frametype == AST_FRAME_VOICE) {
(gdb)
(gdb)
(gdb) bt
# 0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
# 1 0x0000000000416261 in ast_read (chan=0x6439f0) at channel.c:1337
# 2 0x000000000041aa42 in ast_waitfordigit (c=0x6439f0, ms=2) at
channel.c:1140
# 3 0x0000002a9e326af1 in sccp_start_channel (data=0x0) at sccp_pbx.c:505
# 4...
2003 Oct 30
0
SIP error: Asked to transmit frame type 64
...r/${EXTEN:1})
exten => _9XXXX,5,Congestion
exten => _9XXXX,105,Playback(tt-monkeysintro)
exten => _9XXXX,106,Hangup
my chan_sip.c:
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast->pvt->pvt;
int res = 0;
if (frame->frametype == AST_FRAME_VOICE) {
if (!(frame->subclass & ast->nativeformats)) {
--> --> ast_log(LOG_WARNING, "Asked to transmit frame type %d, while
native formats is %d (read/write = %d/%d)\n",
frame->subclass, ast->nativeformats, ast->readformat, ast-
>writeformat);
return -1;...
2003 Jul 08
0
Patch to fix some segfaults in Asterisk
...===================================================================
RCS file: /usr/cvsroot/asterisk/frame.c,v
retrieving revision 1.3
diff -u -r1.3 frame.c
--- frame.c 28 Jun 2003 16:40:02 -0000 1.3
+++ frame.c 8 Jul 2003 10:48:15 -0000
@@ -125,8 +125,10 @@
/* Make frame */
s->f.frametype = AST_FRAME_VOICE;
s->f.subclass = s->format;
- s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
- s->f.offset = AST_FRIENDLY_OFFSET;
+ /*s->f.data = s->framedata + AST_FRIENDLY_OFFSET;*/
+ s->f.data = s->framedata;
+ /*s->f.offset = AST_FRIENDLY_OFFSET;*/
+ s->f.offset = 0;
s->...
2005 Aug 26
0
Broken pipe of stdinpcm on asterisk-ices.xml
...ms = ast_waitfor(chan, -1);
if (ms < 0) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
}
f = ast_read(chan);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
res = write(fds[1], f->data, f->datalen);
if (res < 0) {
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
res = -1;
break;
}
}
}
ast_frfree(f);
}
}
close(fds[1]);
LOCAL_USER_REMOVE(u)...
2009 Oct 05
3
Questions about app_jack.c
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular &
1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to