search for: slin

...hone. I followed the instructions... This shows me its loaded core show translation paths opus --- Translation paths SRC Codec "opus" sample rate 48000 --- opus:48000 To g723:8000 : No Translation Path opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at 8000)->(ulaw at 8000) opus:48000 To alaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at 8000)->(alaw at 8000) opus:48000 To gsm:8000 : (opus at 48000)->(slin at 48000 )->(slin at 8000)->(gsm at 8000) o...

...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 (...

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...

...and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[1...

...13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When I do the same from a call file like: same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\n...

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No...

...DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 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...

...translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw" sample rate 8000 --- alaw To g723 : No Translation Path alaw To gsm : (alaw)->(slin)->(gsm) alaw To ulaw : (alaw)->(ulaw) alaw To g726 : No Translation Path alaw To adpcm : No Translation Path alaw To slin : (alaw)->(slin) alaw To lpc10 : No Translation Path alaw...

Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw Voicemail is recorded in wav|ulaw so there should be native playback to g.71...

...apability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody know why? [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 16 == Registered translator 'lintoilbc' from format slin to ilbc, cost 90 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format gsm to slin, cost 5 == Registered translator 'lintogsm' from format slin t...

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame t...

...ogress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data from MP3Player, I get no sound on my speakers and get the following from 'core show channel PJSIP/xxx': NativeFormats: (slin)WriteFormat: slinReadFormat: slinWriteTranscode: No ReadTranscode: No I have noticed that when I do a UNICAST page and send data from MP3Player, everything works flawlessly and I get the following from 'core show channel MulticastRTP': NativeFormats: (ulaw)WriteFormat: slinReadFormat: slin...

...ering if someone of you have the same thing 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...

...500 > Timer B : 32000 > ToHost : > 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 &g...

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

...av file that plays nicely on my 1.4 server. Here is what the console displays: -- Executing [730 at customers:2] Playback("SIP/user_xxx-00000012", "/home/phones/common/moh/moha/Sovereign") in new stack Unable to find a codec translation path from 0x40004 (ulaw|h261) to 0x40 (slin) Unable to open /home/phones/common/moh/moha/Sovereign (format 0x40004 (ulaw| h261)): No such file or directory ast_streamfile failed on SIP/user_xxx-00000012 for /home/phones/common/moh/moha/Sovereign I was under the impression that I didn't have to do anything to get slin support. when I...

...at 17:23:14, Luca Bertoncello wrote: > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer <peername>" for a phone. > bpi*CLI> sip show peer 0049177xxxxxxx > 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|t > estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk > |silk|silk) That strikes me as somewhat unlikely. > bpi*CLI> sip show peer 0049351xxxxxxx > Codecs : (...

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

...and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail...