search for: slin16

...1][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 native formats are (si...

I thought - what about the software - maybe it needs updated. After doing so I get a list: Audio codecs PCMU (8000 Hz) PCMA (8000 Hz) opus (48000 Hz) L16/16000 (16000 Hz) G.726-32 (8000 Hz) L16/8000 (8000 Hz) speex/16000 (16000 Hz) speex/8000 (8000 Hz) -------------- next part -------------- An HTML attachment was scrubbed... URL:

...#39;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?

...? 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 -...

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000 4000 10000 ulaw - 6001 - 1 4001 2 1 14001 - 32003 - 4001 2001 8001...

...hanges on Asterisk 11 (VM, Cloud or even physical machine). Is it slin?, adding this overhead or there is something I am overlooking?. * * *Asterisk 11.0.1 => core show translation **(in microseconds)* *gsm ulaw alaw g726 adpcm slin lpc10 g729 speex speex16 ilbc g726aal2 g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44 slin48 slin96 slin192* *gsm *- 15000 *15000 *15000 15000 9000 15000 15000 *15000 *23000 15000 15000 17250 17000 15000 23000 17000 17000 17000 17000 17000 17000 17000 *ulaw *15000 - 9150 15000 15000 9000 15000 15000...

Ok - updating the firmware on teh device - factory reset, re-config. Capabilities: us - (g726|slin16|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g726|slin16)/video=(nothing)/text=(nothing), combined - (g726|slin16|ulaw|alaw) Looking much better. Jerry On Wed, Jun 17, 2020 at 4:01 PM Jerry Geis <jerry.geis at gmail.com> wrote: > I thought - what about the software - maybe it needs updated....

...nslation Path alaw To slin : (alaw)->(slin) alaw To lpc10 : No Translation Path alaw To g729 : No Translation Path alaw To speex : (alaw)->(slin)->(speex) alaw To speex16 : (alaw)->(slin)->(slin16)->(speex16) alaw To ilbc : No Translation Path alaw To g726aal2 : No Translation Path alaw To g722 : No Translation Path alaw To slin16 : (alaw)->(slin)->(slin16) alaw To siren7 : No Translation Pat...

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

...ligible audio in both directions. So far, the only other clue I have is this pair of messages upon startup: [Feb 7 18:35:02] WARNING[17950] translate.c: plc_samples 160 format f [Feb 7 18:35:02] VERBOSE[17950] logger.c: == Registered translator 'g722tolin16' from format g722 to slin16, cost 1999 A "core show translation recalc 240" shows that all the necessary translations are available and sufficiently fast: Recalculating Codec Translation (number of sample seconds: 240) Translation times between formats (in microseconds) for one second of data...

...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 -...

I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project. Using slin16 format. 1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is this correct? 2) Is there some place I can find a description of the 12-byte packet header fields? Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.co...

...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* - - - -...

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

...operly. *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 - - 2001...

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 1...

Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it. The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

...: 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). Still on B, my...

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 back