I have an analog voice interfaced with sip server through the codec. How can I have my analog voice interact with the sip server using dtmf tones? asteriks does not recognize my dtmf tones since it does not take the connection through the codec as a sip client.
Cengiz Gündoğ Embedded Software Developer Teknobil Inc, _______________________________________________ ekiga-list mailing list [email protected] http://mail.gnome.org/mailman/listinfo/ekiga-list
