Re: [asterisk-users] Sounds format: GSM to G.729
On Fri, Jun 26, 2009 at 4:21 PM, Kevin P. Fleming wrote: > Alejandro Cabrera Obed wrote: > > > Because sounds files in /var/lib/asterisk/sounds are a lot as I see. > > If you are using the Spanish sounds distributed by Digium, they are > already available in G.729 format from downloads.asterisk.org. > > Thanks Kevin, so If I use G.729 in sound files, IP phones and Asterisk and I not need any trascoding to the PSTN, can I use the codec for free absolutely ??? Thanks again. Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sounds format: GSM to G.729
Alejandro Cabrera Obed wrote: > Because sounds files in /var/lib/asterisk/sounds are a lot as I see. If you are using the Spanish sounds distributed by Digium, they are already available in G.729 format from downloads.asterisk.org. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sounds format: GSM to G.729
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 lot Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
