search for: frametype

...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...

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...

...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_sc...

...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 enter...

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 &quo...

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<\0...

...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) S...

...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:...

...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][...

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...

...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) at...

...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 = (st...

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

...-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- >writeform...

...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 */ } ---...

...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 co...

..._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 idea...

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

...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/

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