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Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I trascode sounds file from GSM to G.729 ??? Any Linux package suggestion to do this task ??? Because sounds files in /var/lib/asterisk/sounds are a lot as I see. Thanks a...

...license, using Asterisk in "pass-through" mode for G.729 voice communications, do I just have to download the open source version of the G.729 codec or can I use the one coming in Asterisk ??? 3) If I use G.729 for voice communications and GSM for voice mail sounds, does Asterisk execute trascoding ??? Really thanks Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090721/93d64237/attachment.htm

...10:58 AM, Alejandro wrote: > When you say as Source Client, means like listener? > > i'm using for push and pull the streaming. > > For push, we capture from SDR Dongle and LineIn audio card, encode and > push to Icecast server with FFMPEG. > > And to play (and process, trascoding too) to multiple bitrate and > others task too. > > What is your doubt? > > Regards > Alejandro > > El mar., 27 de ago. de 2019 a la(s) 12:33, Jack Elliott > (that.jack.elliott at gmail.com <mailto:that.jack.elliott at gmail.com>) > escribió: > >...

As described here in ffmpeg's documentation: http://ffmpeg.org/ffmpeg-all.html#Icecast -- That Jack Elliott (541) 848 7021 KPOV 88.9 FM High Desert Community radio Host, The Sunday Classics

Hi all - We are using voicepulse connect & asterisk together. We'd like to record our own outgoing messages, to be played back people that will be dialing into our voicepulse connect supplied DID. (am I getting this lingo right?) What is the best audio format to ensure the highest level of audio quality? Thank you - Matt Pease Parking Hero

Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? Thanks Ish --