Le mercredi 18 février 2009 à 09:19 +0200, [email protected] a écrit : > 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.
Asterisk is supposed to do that for you : convert analog DTMFs to SIP DTMFs and vice-versa. I propose you ask your questions on their mailing list. -- _ Damien Sandras (o- //\ Ekiga Softphone : http://www.ekiga.org/ v_/_ Be IP : http://www.beip.be/ FOSDEM : http://www.fosdem.org/ SIP Phone : sip:[email protected] _______________________________________________ ekiga-list mailing list [email protected] http://mail.gnome.org/mailman/listinfo/ekiga-list
