It's more than highly likely you have some other problem like jitter or a bad network connection. Not many people would be able to tell the difference between the sound of an 8k PCM file and the same file encoded to G711 just by listening to it unless there was a severe problem somewhere. Since you are behind NAT you are even more likely to experience drops etc.
Record your files as 8k raw 16 bit PCM to get the best out of the file playback in FS and look elsewhere for your audio issues. You can always make sure you are using the latest build of FS to rule out any temporary issues in the code. On Wed, Dec 23, 2009 at 8:56 AM, Brian West <[email protected]> wrote: > VMD will force a transcode anyway too. > > /b > > On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > > > My setup is as follows: > > > > FreeSWITCH -> SIP Trunk -> PSTN. > > > > From freeswitch, I'm making outbound calls using event socket via the > "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is > default settings. Using "playback" application, I'm playing a mu-law audio. > I'm also starting the "vmd" application, so that I can replay the message on > beep. > > > > Thanks for your suggestion on native format. I'll try it. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:+19193869900
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