Surprisingly, despite the error message, the files is uploaded in "/var/lib/asterisk/mohmp3" with correct permissions and ownership. Its not showing in FreePBX MOH Screen. I guess its a FreePBX issue.
Sans On Thu, Sep 1, 2011 at 7:56 PM, RSCL Mumbai <rscl.mum...@gmail.com> wrote: > Thanks again @Danny. > > File converter worked like a charm. > asterisk -rx "file convert /var/lib/asterisk/mohmp3/wav_Track11.wav > wav_Track11.alaw" > > I coped the new file from sounds/ folder to my desktop > And I tried to upload the new .alaw file using FreePBX, > > I got the following error: > > Error Processing: "sox failed to convert file and original could not > be copied as a fall back" for wav_Track111.alaw! > This is not a fatal error, your Music on Hold may still work. > > > Pls help with this last bit. > > Thx > Sans > > > > > On Thu, Sep 1, 2011 at 7:17 PM, Danny Nicholas <da...@debsinc.com> wrote: >> Asterisk has a built-in file convert >> >> help file convert >> Usage: file convert <file_in> <file_out> >> Convert from file_in to file_out. If an absolute path is not given, the >> default Asterisk sounds directory will be used. >> >> Example: >> file convert tt-weasels.gsm tt-weasels.ulaw >> >> -----Original Message----- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai >> Sent: Thursday, September 01, 2011 8:26 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] cli command show codecs >> >> Thx @Danny >> >> I am feeling a bit lost here... >> >> We are using G711-aLaw for all our calls (endpoints) and I would like to >> align everything to this codec. >> >> I have an MOH file -- a custom wav file. How do I check its codec format ? >> >> And if its not G711-aLaw, how do I convert it to G711-aLaw. >> >> Thank you. >> Sans >> >> >> >> >> >> On Thu, Sep 1, 2011 at 6:35 PM, Danny Nicholas <da...@debsinc.com> wrote: >>> Maybe this will be better than my first answer – Audio files do indeed >>> have codec formats. If you are in a particular codec (say G729), >>> Playback/Background and MOH will search for files that match the codec >>> format first, then transcode WAV/GSM/whatever you have to that format >>> if it isn’t found. Ideally, you want to have a copy of each codec you >>> can play for all sounds and MOH. Each of the “canned sounds” comes in >>> each codec format (you pick the ones you want when you do make menuselect). >>> >>> >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> Sent: Thursday, September 01, 2011 5:35 AM >>> >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] cli command show codecs >>> >>> >>> >>> Hi, >>> >>> Does audio files have codec formats? I simply convert all my audios >>> (MOH, >>> accouncements) to .wav format, 16bit, 11kHz (I believe this is the >>> best format for asterisk). >>> I am new to this and may be incorrect. >>> >>> Going forward, >>> (a) How can I check the codec format of my announcements, MOH ? >>> (b) How can I record/convert announcements, MoH etc to a particular format ? >>> >>> I believe its a good idea to prevent transcoding and save CPU overheads. >>> >>> Thx >>> Sans >>> >>> >>> On Thu, Sep 1, 2011 at 11:39 AM, Bruce B <bruceb...@gmail.com> wrote: >>> >>> if you see leg-A as ulaw and leg-B as g729 then it's trans-coding. If >>> your IVR announcement is not recorded in g729 and you see g729 on the >>> channel when you call into IVR then it's transcoding as well. >>> >>> >>> >>> On Wed, Aug 31, 2011 at 5:50 PM, Eric Wieling <ewiel...@nyigc.com> wrote: >>> >>> Assuming SIP "sip show channels" will show you which codec is used for >>> each call leg. However it does not track transcoding. >>> >>> -----Original Message----- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> >>> Sent: Wednesday, August 31, 2011 2:45 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> >>> Subject: Re: [asterisk-users] cli command show codecs >>> >>> asterisk -rx "core show channels verbose" does not provide transcoding >>> details. >>> >>> Unless I have missed something. >>> >>> Sans >>> >>> >>> >>> On Wed, Aug 31, 2011 at 10:34 PM, Danny Nicholas <da...@debsinc.com> wrote: >>> >>> >>> Core show channels verbose is probably your best bet. I think >>> the answer also depends on your * version. >>> >>> >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL >>> Mumbai >>> Sent: Wednesday, August 31, 2011 10:44 AM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [asterisk-users] cli command show codecs >>> >>> >>> >>> Hi, >>> >>> Is there a CLI command which will tell me the codec used for >>> active calls and if transcoding is happening ? >>> >>> Thx >>> Sans >>> >>> >>> -- >>> >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by >>> http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >> Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users