search for: siren7

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)? Thank...

...07:24:55] WARNING[23961][C-00000000]: translate.c:490 ast_translator_build_path: No translator path: (starting codec is not valid) [Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw frame when native formats are (siren7) (rd:alaw->slin16;(alaw at 8000)->(slin at 8000)->(slin at 16000) wr:slin16->alaw;(slin at 16000)->(slin at 8000)->(alaw at 8000)) [Oct  2 07:24:55] WARNING[23961][C-00000000]: chan_pjsip.c:856 chan_pjsip_write: Channel PJSIP/boslwzldi21-00000001 asked to send alaw frame when...

...ren14.48kbps Siren22.64kbps G7221.16kbps G7221_C.24kbps G7221.24kbps G7221_C.32kbps G7221.32kbps G7221_C.48kbps I can get Asterisk to work with G722 and the sound is superior compared to uLAW. I tried to get it working with Siren7 and Siren14 but I cannot. It always says incompatible codec and what is this Siren14 and Siren22 with Polycom. Is this different to Asterisk's Siren7 and Siren14? Is this why I cannot get Siren to function? siren7 show version : Digium Siren7 Module Version 1.8.0_1.0.5 (optimized for...

What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work?

> 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).

...g on CentOS 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...

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 =======================================

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

...ng. Doing 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...

...*> 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* - - - - - 2000 ulaw...

...d install 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...

...10) (0x400) audio ilbc (iLBC) 2048 (1 << 11) (0x800) audio 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))...

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

...ost : 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 Mi...

...gt; (iLBC) >> 2048 (1 << 11) (0x800) audio 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...

...00 14500 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...

...To g729:8000 : No Translation Path speex:8000 To speex:16000 : No Translation Path speex:8000 To speex:32000 : No Translation Path speex:8000 To ilbc:8000 : No Translation Path speex:8000 To g722:16000 : No Translation Path speex:8000 To siren7:16000 : No Translation Path speex:8000 To siren14:32000 : No Translation Path speex:8000 To testlaw:8000 : No Translation Path speex:8000 To g719:48000 : No Translation Path speex:8000 To opus:48000 : No Translation Path speex:8000 To none:8000...

...: (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...

...00) audio ilbc > (iLBC) > 2048 (1 << 11) (0x800) audio 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 Linea...

...Encryption : No Callerid : "" <> Expire : -1 ACL : Yes Addr->IP : 10.215.147.115 Port 4569 Defaddr->IP : (null) Port (null) Username : interbox Codecs : (g723|gsm|ulaw|alaw|g726aal2|adpcm|slin|lpc10|g729|speex|ilbc|g726|g722|siren7|siren14|slin16|h264|vp8|opus) Codec Order : (ulaw|alaw|siren14|siren7|g722|slin16|slin|g726|g726aal2|adpcm|gsm|ilbc|speex|lpc10|g729|g723|opus|vp8|h264) Status : OK (3 ms) Qualify : every 60000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off) Asterisk ending (0)....