<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">You could also use the inband dtmf generator and queue_dtmf.<div><br></div><div>Mike</div><div><br><div><div>On Jul 15, 2008, at 9:45 AM, Anthony Minessale wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr">there is no way to do both inband and info/2833 from within sip.<br><br>inband dtmf is not part of the SIP module it's part of the core.<br><br>you can generate an inband dtmf stream with gentones app or the tone_stream:// file stream.<br> <br><action application="gentones" data="1234567890"/><br>or<br><action application="playback" data="tone_stream://1234567890"/><br><br><div class="gmail_quote">On Tue, Jul 15, 2008 at 3:29 AM, John Wehle <<a href="mailto:john@feith.com">john@feith.com</a>> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">> Try a sleep after the answer.<br> <br> Will do ... just curious as to why and for how long?<br> <br> >> 1) When I dial the extension from a Grandstream GXP-2000 ...<br> >> How do I configure FreeSWITCH to send both RTP digits and inband audio?<br> > The VoIP phone has no reason to reproduce any DTMF.<br> <br> I understand that the VoIP phone has no reason to create DTMF tones from<br> the RTP digits sent by FreeSWITCH. What I was asking is how to you<br> configure FreeSWITCH to generate inband DTMF tones in addition to<br> the RTP digits.<br> <br> I.e. this particular phone can be configured to send DTMF digits to<br> FreeSWITCH as any combination of inband audio, RTP digits, and / or<br> SIP info. Going in the opposite direction I see where in the FreeSWITCH<br> sofia xml configuration type you can set the dtmf-type as either rfc2833<br> or as SIP info however I don't see an option for using both nor do I see<br> an option for generating inband DTMF audio.<br> <br> Note: I was just using send_dtmf with the VoIP phone for testing purposes<br> ... this application doesn't actually need to send DTMF to a VoIP phone.<br> <br> >> 2) When I dial the extension from a FXO port on a Sangoma A204D it connects,<br> > Is this using OpenZAP?<br> <br> Yes. In my particular application the System 25 PBX connects to FreeSWITCH<br> using OpenZAP running on a Sangoma A204DX card. At the end of a call sent<br> to voicemail I need FreeSWITCH to send some DTMF tones to the PBX in order<br> to turn on / off the message waiting indicator.<br> <br> -- John<br> -------------------------------------------------------------------------<br> | Feith Systems | Voice: 1-215-646-8000 | Email: <a href="mailto:john@feith.com">john@feith.com</a> |<br> | John Wehle | Fax: 1-215-540-5495 | |<br> -------------------------------------------------------------------------<br> <br> <br> _______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> </blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br> <br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br> IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400 </div> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>