This happens here, it is due to the lack of jitter buffer in the sip channel, and using ulaw as the codec. Switch to GSM if you can, or wait for the sip jitter buffer to be completed...
On Wed, 2005-23-03 at 15:29 -0800, Sean Kennedy wrote: > Pol wrote: > > > I'm using asterisk as a sip client with a sip proxy server... I've > > made the pertinent extensions and I've configured the sip.conf > > correctly or I think so.. > > > > I'm using x-lite as a client and when I ring to a public telephone > > through proxy, the arriving sound it's perfect but the sound I send is > > very bad, they hear me like a robot and distorted. > > > > Anyone know what's the problem? > > > > Thank you very much. > > > > Pol. > > What codecs are you using? Between xlite and asterisk, and asterisk and > the sip server? > > Sean > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Scott J. Williamson <[EMAIL PROTECTED]> tmps_base = tmps_max; /bin /boot /cdrom /dev /etc /home /initrd /initrd.img /initrd.img.old /lib /lost+found /media /mnt /opt /proc /root /sbin /srv /sys /tmp /usr /var /vmlinuz /vmlinuz.old protect our mortal string backups/ bin/ commapi/ COUT/ cvsroot/ Desktop/ dragoneye/ eagle/ eggdrop/ freenet/ hwdev/ i2p/ laptop backup/ Mail/ mnt/ mutella/ public_html/ src/ Templates/ tp/ unnamed/ -- Larry Wall in stab.c from the perl source code _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
