Displaying 20 results from an estimated 30 matches for "frametyp".
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frametype
2015 Aug 12
1
enableJIT in Rprofile leads to 'not a proper evaluation environment' on startup
...tart------------->8---
library(compiler)
compiler::enableJIT(3)
--8<---------------cut here---------------end--------------->8---
in ~/.Rprofile for years; now that I upgraded to 3.2.1 I get this on startup:
--8<---------------cut here---------------start------------->8---
Error in frameTypes(env) : not a proper evaluation environment
Calls: <Anonymous> -> <Anonymous> -> makeCenv -> structure -> frameTypes
> q()
Warning message:
restarting interrupted promise evaluation
--8<---------------cut here---------------end--------------->8---
I don't see...
2004 Oct 05
1
Brazillian Caller ID: almost there...
Hello,
Talking with Soren Sratje about Caller ID in Brazil, we compare ours
DTMF tones captured by 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...
2005 Dec 28
5
Regular crashes
...ds
Andrew Gough
FIRST TRACE
#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at lock.h:597
No locals.
#2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at channel.c:671
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec =
0,
tv_usec = 0}, prev = 0x0, next = 0x0}
#3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at chan_sip.c:1315
p = (struct sip_pvt *) 0x81be208
#4 0x08056c3e in ast_s...
2003 Oct 12
2
INFO method and DTMF translation
...e if (event < 12) {
resp = '#';
} else if (event < 16) {
resp = 'A' + (event - 12);
}
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
f.subclass = resp;
f.offset = 0;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(p->owner, &f, 0);
}
On line 3986, any # or * digit I ente...
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 &qu...
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<\...
2005 Aug 01
0
Music on hold problem.
...3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
RFC3389: 1 bytes, level 8...
channel debug:
RFC3389: 1 bytes, level 8...
<< [ TYPE: Unknown Frametype '10' (10) SUBCLASS: Unknown Subclass (64)
] [SIP/5060-b5300b98]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/5060-b5300b98]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/5060-b5300b98]
RFC3389: 1 bytes, level 8...
<< [ TYPE: Unknown Frametype '10' (10)...
2006 Oct 16
1
Page hangs up after 5 seconds
...rom unknown to ulaw
Oct 16 11:01:12 DEBUG[6767] rtp.c: RTCP NAT: Got RTCP from other end.
Now sending to address 212.247.4.149:49435
Oct 16 11:01:12 DEBUG[6767] rtp.c: Got RTCP report of 52 bytes
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Got unrecognized frame on
channel SIP/snom1-b7d0c328, f->frametype=5,f->subclass=0
Oct 16 11:01:12 DEBUG[6767] rtp.c: RTP NAT: Got audio from other end.
Now sending to address 212.247.4.149:49434
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Got unrecognized frame on
channel SIP/snom1-b7d0c328, f->frametype=5,f->subclass=0
Oct 16 11:01:12 DEBUG[6771] rtp.c...
2014 Oct 23
1
Auto video call hangup
...C-00000012] res_rtp_asterisk.c: Created
smoother: format: ulaw ms: 20 len: 160
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Starting
RTCP transmission on RTP instance '0xb69b9894'
[Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Packet arrived out of order
(expecting 7, got 5) (frametype = 3, subclass = 200004)
[Oct 24 16:33:49] DEBUG[15204] chan_iax2.c: Acking anyway
[Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Packet arrived out of order
(expecting 7, got 6) (frametype = 2, subclass = 100003)
[Oct 24 16:33:49] DEBUG[15211] chan_iax2.c: Acking anyway
[Oct 24 16:33:49] DEBUG[15590]...
2017 Apr 12
2
More issues with Siren14 datalen == 0 packets
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
uint32 = 2156475160, pad = "\030\06...
2005 Jan 31
1
chan_sccp bug / problem
...64/libnsl.so.1
Reading symbols from /usr/lib/asterisk/modules/chan_sccp.so...done.
Loaded symbols for /usr/lib/asterisk/modules/chan_sccp.so
Reading symbols from /lib64/libgcc_s.so.1...done.
Loaded symbols for /lib64/libgcc_s.so.1
#0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
38 if (f->frametype == AST_FRAME_VOICE) {
(gdb)
(gdb)
(gdb) bt
# 0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
# 1 0x0000000000416261 in ast_read (chan=0x6439f0) at channel.c:1337
# 2 0x000000000041aa42 in ast_waitfordigit (c=0x6439f0, ms=2) at
channel.c:1140
# 3 0x0000002a9e326af1 in sccp_start_channel (data=0x0) a...
2006 Jan 05
0
Regular Crashes - Partially Solved
...thread_mutex_trylock () from
/lib/tls/libpthread.so.0
> > No symbol table info available.
> > #1 0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at
lock.h:597
> > No locals.
> > #2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at
channel.c:671
> > f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
> > mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec
=
> > 0,
> > tv_usec = 0}, prev = 0x0, next = 0x0}
> > #3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at
chan_sip.c:1315
> > p = (s...
2003 Jun 17
1
i4l - summary of patches?
Hi,
I'm trying to get asterisk running on kernel 2.4.20 however trawling through
the archives I've found a few references to patches to remove i4l's dtmf
detection, but have been unable to find the patch itself (I think it is
isdn_audio.c). Can anyone point me in the right direction?
The problem I'm seeing is connecting a SIP softphone (tried a few) to an
external number via an
2003 Oct 30
0
SIP error: Asked to transmit frame type 64
...-regist-server/${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-
>writefor...
2003 Dec 03
0
Implement missing features in Meetme application
...mp; user list ),
I'm starting to implement the missing features in Meetme application :
's' -- send user to admin/user menu if '*' is received
Line 438
-------- app_meetme.c -----------------------------------------------------------------------------
else if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*') && (confflags & CONFFLAG_STARMENU)) {
if ((confflags & CONFFLAG_ADMIN)) {
/* Do admin stuff here */
} else {
/* Do user menu here */
}
--...
2003 Dec 05
0
Native bridging with Polycom 600
...y sip.conf has
canreinvite=yes for both phones. They connect, and I can talk as usual, but
sniffing shows the RTP stream is routed through Asterisk.
The exact spot where the attempt to natively bridge fails is in rtp.c, line
1281 (CVS from October 8, 2003):
f = ast_read(who);
if (!f || ((f->frametype == AST_FRAME_DTMF) &&
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))))
A bit of logging shows the frame f is NULL, so Asterisk thinks one side has
hung up, and gives up on the bridging attempt. Of c...
2005 Sep 09
0
Doesn't finishes callerid spill
..._log(LOG_WARNING, "Didn't finish Caller-ID spill. Cancelling.\n");
free(p->cidspill);
p->cidspill = NULL;
p->callwaitcas = 0;
}
p->subs[index].f.frametype = AST_FRAME_CONTROL;
p->subs[index].f.subclass = AST_CONTROL_RINGING;
break;
***********************************************************************
I am seaching Why loop exits before reaching limit of 8867 or what
makes zt_handle_event to control the flow.
Please help me with any ide...
2007 May 25
1
H Parameter in Dial Command
Hi List,
I am currently using the H parameter in the dial command. The issue that I am having is that if the user is calling an ivr that requires him to press the * key then the call gets hung up on. How would I go about changing it so that the user will have to press say ** for the H parameter to come in to effect ?
Thanks a lot.
Dovid
-------------- next part --------------
An HTML attachment
2009 Feb 04
0
Stopping chanspy
...g == 99999) running = -1;
break;
Then hitting 99999# on the keypad does what I want.
I do not particularly want to patch future versions of the code to add this
ability.
The only other conditions that set running to -1 is if you hang up the
channel or res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0; sets
res < 0. As I do not know what I can type on the keypad to cause that I do
not know how to exit the chanspy application.
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
http://www.cfmc.com/
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to