if you make a dialplan that plays a file but does not answer then it will happen in the early media stage. DTMF working during the call is dependent on the other side. We certainly support it but some carriers purposely do not allow DTMF during progress since running an ivr in progress stage means you are not paying them any money for the calls.
On Mon, Jan 5, 2009 at 7:27 AM, shehzad p <[email protected]> wrote: > > Hi All, > Is there a way in freeswitch, such that we can play sound and receive a > DTMF > from the end user without actually answering a call like in session > progress > stage. > Basically, the system should accept the call and play a progress tone > during > call progress (SIP 183). The system should be able to collect DTMF input. > All this should happen while in SIP 183 (Progress) and the call shouldn't > change its state to CONNECT, which means, no 200 OK should be issued to the > use via the system. This is more like a IVR based system, however it all > happens during call progress and not after connect. > Can anyone suggest any idea about this situation? > > Thanks in advance. > -- > View this message in context: > http://www.nabble.com/Idea-Suggestion-required-tp21291002p21291002.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > 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] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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