search for: ast_frame

...| | | If the frame is going to be transmitted later, an | | | iax_frame struct is dynamically allocated with a data | | | buffer that has the exact buffer size needed to | | | accommodate for the provided ast_frame data. However, if | | | the frame is being sent immediately, it uses a stack | | | allocated iax_frame, with a data buffer size of 4096 | | | bytes. | | |...

...| | | If the frame is going to be transmitted later, an | | | iax_frame struct is dynamically allocated with a data | | | buffer that has the exact buffer size needed to | | | accommodate for the provided ast_frame data. However, if | | | the frame is being sent immediately, it uses a stack | | | allocated iax_frame, with a data buffer size of 4096 | | | bytes. | | |...

...you haven't done so already, would be to > try it without MeetMe: > > Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE -> > radio gateway > > If that works, then it would suggest that the SIP MESSAGE is > successfully getting translated into an ast_frame, which is then getting > translated back into a SIP MESSAGE. If that is not happening, you might > need to add some code to chan_sip.c to do those steps. > > Once Asterisk is converting the message to and from an ast_frame, the > next step would be to add some code to app_meetme.c in...

Hi, if I dial remote user (over CAPI) and transfer it to conference, if it presses * or #, asterisk crashes with following core trace... This works OK with internal extensions .... bt full 0 0x0805dae6 in ast_queue_frame (chan=0x819b738, fin=0x40d99660) at channel.c:384 f = (struct ast_frame *) 0x810d7b8 prev = (struct ast_frame *) 0x0 cur = Variable "cur" is not available. I have : exten => 80,1,MeetMe(|aMsE) exten => _8[1-8],1,MeetMe(${EXTEN}|pMs) I've submitted bug report, but they claim I should recreate it on official Asterisk channel. Anyon...

...running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to p->obuflen which is zero. Has anybody noticed this, too? Since I don't know the code, I cannot suggest a fix, but maybe someone who does is listening? :) For testing purposes, I changed the source line to if (...

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

..., we need to just > > // write the SLINEAR frame. > > if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) { > > struct ast_trans_pvt* transPath = > ast_translator_build_path(chan->writeformat, AST_FORMAT_SLINEAR); > > struct ast_frame* transFrame = ast_translate(transPath, > &ps->f, 0); > > if (transFrame) { > > ast_write(chan, transFrame); > > ast_frfree(transFrame); > > } > > ast_translator_free_path(t...

...XX,2,Ringing exten => _9XXXX,3,Wait(1) exten => _9XXXX,4,Dial(IAX2/myserv:mypw@remote-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&q...

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310 int ms; [...] 2365 /* backward compatibility, if no offset given, arg[6] would have been 2366 * caught below and taken to be a beep, else if it is a digit then it is...

...=================================================================== 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;...

...art of "app_ices.c" is as follows: --------------------------------------------------------- static int ices_exec(struct ast_channel *chan, void *data) { int res=0; struct localuser *u; int fds[2]; int ms = -1; int pid = -1; int flags; int oreadformat; struct timeval last; struct ast_frame *f; char filename[256]=""; char *c; last.tv_usec = 0; last.tv_sec = 0; ...(snip)..... res = icesencode(filename, fds[0]); close(fds[0]); if (res >= 0) { pid = res; for (;;) { /* Wait for audio, and stream */ ms = ast_waitfor(chan, -1); if (ms < 0) { ast_log(L...

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

...o idea what the problem is. Anyone can give me some advices. --core dump 1-- (gdb) bt full #0 0x00000037e806e1f3 in _int_free () from /lib64/libc.so.6 No symbol table info available. #1 0x00000037e8071fac in free () from /lib64/libc.so.6 No symbol table info available. #2 0x000000000046b7b7 in ast_frame_free (fr=0x1b9da4b0, cache=0) at frame.c:369 No locals. #3 0x00002aaab1173573 in mixmonitor_thread (obj=0x1bb08220) from /usr/lib/asterisk/modules/app_mixmonitor.so next = (struct ast_frame *) 0x0 write = 1 mixmonitor = (struct mixmonitor *) 0x1bb08220 f = (s...

...ssary. > > I'm revisiting this issue now. It looks like it would help to have the ability to: 1) Take a look at some speex data, and return the number of samples it contains. This would go here in asterisk, for example: asterisk/channels/chan_iax2.c: static int get_samples(struct ast_frame *f) { int samples=0; switch(f->subclass) { case AST_FORMAT_SPEEX: samples = 160; /* XXX Not necessarily true XXX */ break; case AST_FORMAT_G723_1: samples = 240 /* XXX Not necessarily true XXX */; br...

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

...media streams results in 488 (Reported by Matt Jordan) * ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell) * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fix...

...media streams results in 488 (Reported by Matt Jordan) * ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell) * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fix...

Jean-Marc Valin wrote: >>Heh. I guess after playing with different jitter buffers long enough, >>I've realized that there's always situations that you haven't properly >>accounted for when designing one. >> >> > >For example? :-) > > I have a bunch of examples listed on the wiki page where I had written initial specifications:

...ocks with DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan) * ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell) * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined media streams results in 488 (Reported by Matt Jordan) * A...