Asterisk maybe? > -----Original Message----- > From: Gene Cohen [mailto:[EMAIL PROTECTED] > Sent: Wednesday, June 07, 2006 2:46 PM > To: [email protected] > Subject: [Users] Call Flow > > > > I am developing an application which requires the following call flow: > > 1. SIP Phone makes call which arrives at openser > 2. Before processing the call openser connects the call to > another address > where the user hears a recorded message > 3. When that call ends I want to connect the original SIP > call as requested. > > Has anyone done anything like this before? > > Thanks > gene > > > > _______________________________________________ > Users mailing list > [email protected] > http://openser.org/cgi-bin/mailman/listinfo/users >
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