Asterisk maybe?
-----Original Message----- From: Gene Cohen [mailto:gcohen@excelity.com] Sent: Wednesday, June 07, 2006 2:46 PM To: users@openser.org Subject: [Users] Call Flow
I am developing an application which requires the following call flow:
- SIP Phone makes call which arrives at openser
- 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 Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users