problem is if this is over sip, the dtmf is queued into the rtp stack and sent in real time as the call progresses. Since you are hanging up right away, it never gets a chance to send it. you must sleep long enough for the entire tone to be sent.
I added a convicence variable set once you call send_dtmf that will tell you how long to sleep until all of the digits should be sent. update to latest trunk and try this: JavaSession s = new JavaSession(uuid); s.answer(); s.streamFile("/usr/local/freeswitch/sounds/1.wav"); s.execute("send_dtmf", "[EMAIL PROTECTED]"); s.execute("sleep", s.getVariable("last_dtmf_duration")); s.hangup(); On Thu, Sep 11, 2008 at 11:43 AM, Klaus Teller <[EMAIL PROTECTED]> wrote: > HI Brian, > > Thanks for your suggestion. I just don't see how this would help me. I > understand queue_dtmf is to be used before bridging. But i'm not bridging > calls. I'm just originating calls and interacting with the remote device. > What i really want is the inverse of getDigits() that you can call anytime > in the call to send DTMFs. The Asterisk equivalent to what i need would be > SendDTMF. > > Any further idea? > > Klaus. > > > > -------- Original-Nachricht -------- > > Datum: Thu, 11 Sep 2008 11:11:17 -0500 > > Von: Brian West <[EMAIL PROTECTED]> > > An: freeswitch-users@lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] DTMF Reading and Playing > > > Might want to try queue_dtmf > > > > /b > > > > On Sep 11, 2008, at 11:03 AM, James Green wrote: > > > > > Klaus Teller wrote: > > >> I tried the following but for unknown reason, the caller is not > > >> getting anything: > > >> > > >> > > >> JavaSession s = new JavaSession(uuid); > > >> s.answer(); > > >> s.streamFile("/usr/local/freeswitch/sounds/1.wav"); > > >> s.execute("send_dtmf", "[EMAIL PROTECTED]"); > > >> s.hangup(); > > >> > > >> > > >> I can play the file 1.wav without problem but the "send_dtmf" is > > >> simply being ignored. I used wireshrack to check if maybe the > > >> outbound DTMF was sent and not played by my softphone. But this is > > >> not the case. > > > > > > streamFile() blocks until sound file ends or a DTMF tone is > > > received, as > > > detailed on the wiki: > > > > > > http://wiki.freeswitch.org/wiki/Session_streamFile > > > > > > I suspect you want some background music? I'm still trying to get my > > > head around which programming features to use in which circumstances, > > > something I've not found any clear high level guide on yet. > > > <james_green.vcf>_______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users@lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/frees > witch-users > > > http://www.freeswitch.org > > > > Brian West > > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > > > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > 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/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] pstn:213-799-1400
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