Just for integration, look here http://lists.digium.com/pipermail/asterisk-users/2003-July/016464.html
basically sip info dtmf are: Event encoding (decimal) _________________________ 0--9 0--9 * 10 # 11 A--D 12--15 Flash 16 matteo Il dom, 2003-10-12 alle 09:38, Nguyen Hoang Lan ha scritto: > Hello guys, > I have searched high and low, but not found any information about > rules of using DTMF in SIP INFO method. Cisco has described something with > Signal=, but it look like this feature is dependent on implementors? > > The problem is chan_sip.c cannot correctly translate received DTMF > digits, especially #,*. At least with my Antek EGW-804 gateway. > > Looking into chan_sip.c, I found this code: > > line 3982 > if (p->owner) { > if (strlen(buf)) { > if (sipdebug) > ast_verbose("DTMF received: '%c'\n", buf[0]); > event = atoi(buf); << WHY? > if (event < 10) { > resp = '0' + event; > } else if (event < 11) { > resp = '*'; > } else if (event < 12) { > resp = '#'; > } else if (event < 16) { > resp = 'A' + (event - 12); > } > memset(&f, 0, sizeof(f)); > f.frametype = AST_FRAME_DTMF; > f.subclass = resp; > f.offset = 0; > f.data = NULL; > f.datalen = 0; > ast_queue_frame(p->owner, &f, 0); > } > > On line 3986, any # or * digit I entered was translated to 0(zero). So > any apps depends on # for terminating (voicemail for example) won't > work. > > My question is , why not take just buf[0]? why translate? my UA > always send something like d= (one digit) at a time. -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users