search for: siren14

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all code...

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks ----------...

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps G7221_C.24kbps G7221.24kbps G7221_C.32kbps G7221.32kbps G7221_C.48kbps I can get Asterisk to wor...

There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play...

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16?

I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14. I downloaded the codecs and now it will properly transcode to connect to other phones and play any files that are in .wav format. But when it tries to play any files with .siren14 extensions, I get complete noise coming out. Here's the negotiated SDP: v=0 o=root 1668560220 1668560220 IN IP4...

...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"}, delivery = { tv_sec = 1491485582, tv_usec = 407272}, frame_list = {next = 0x0}, flags = 0, ts = 0, len = 0, seqno = 0} frame->data.ptr is an out-of-range address. Does this ri...

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\000"}, delivery = { tv_sec = 1492000520, tv_usec = 225198}, frame_list = {next = 0x0}, flags = 0, ts = 0, len = 0, seqno = 0} Note that datalen is zero, but samples aren't....

...ntOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000 6999 - - 13998 ulaw - 5000 - 1 4000 2 1 8000...

> All patches need to go into JIRA with a license agreement to be > accepted. Understood, but I was using it as an illustration. Note, however, that, from a legal perspective, a patch such as this has no protectable IP (you can't copyright the only way of doing something) and the GNU projects have a formal rule that sufficiently-small patches need no assignments for that reason, which

> A Siren codec is not currently available and the one for 12 will not > work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why).

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

...ing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 2 2 4001 2 1 2 - - - 4001 4002 - - 4003 ulaw - 4001 - 1 4001 2 1 2...

...iled and load in asterisk. *> core show translation * Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr siren14 slin16 g719 speex16 siren7 testlaw g723 - - - - - - - - - - - - - - - - - - - - gsm - - 1001 1001 - - 1000 - - 10999 - - - *9998* - - -...

...ll it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 2 2 2000 2 1 3001 3000 - - 2001 1001 - - 2001 ulaw - 3000 - 1 2000 2 1 3001 3000 - -...

...g726 (G.726 RFC3551) 4096 (1 << 12) (0x1000) audio g722 (G722) 8192 (1 << 13) (0x2000) audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 16384 (1 << 14) (0x4000) audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 32768 (1 << 15) (0x8000) audio slin16 (16 bit Signed Linear PCM (16kHz)) 65536 (1 << 16) (0x10000) image jpeg (JPEG image) 131072 (1 << 17)...

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

...: Addr->IP : (null) Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) Auto-Framing : No Status : UNKNOWN Useragent : Reg. Contact : Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess...

...4096 (1 << 12) (0x1000) audio g722 >> (G722) >> 8192 (1 << 13) (0x2000) audio siren7 (ITU >> G.722.1 (Siren7, licensed from Polycom)) >> 16384 (1 << 14) (0x4000) audio siren14 (ITU >> G.722.1 Annex C, (Siren14, licensed from Polycom)) >> 32768 (1 << 15) (0x8000) audio slin16 (16 >> bit Signed Linear PCM (16kHz)) >> 65536 (1 << 16) (0x10000) image jpeg (JPEG >>...

...0 14500 8500 14500 14500 14500 14500 14500 14500 14500 8500 14500 14500 8500 8500 8500 8500 8500 8500 - *Asterisk 1.6.2.x => core show translation **(in microseconds)* *g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16* *g723* - - - - - - - - - - - - - - - - *gsm *- - 2 *2 *4001 2 1 4001 4001 12002 8002 *4002 *2 - - 2 *ulaw *- 2 - 1 4001 2 1...