Hi! I'm working on a project of integration radios into VoIP network. Main task is SIP to Radio intercom using multicast streams. I already have a working solution, based on Cisco E&M and twisted pair WAVE. Since Wave is a little bit costly solution, I want to implement intercom using freeswitch. Main problem is that I'm not developer and just a network guy and I need help in implementation. I had some experience with multicast steams on Asterisk and I believe that freeswitch is more flexible than asterisk. First and most important question is about how is multicast implemented on freeswitch. Here is an example from the samples included with freeswitch: <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> expression="^pagegroup$|^7243"> - what does this expression mean? Does it mean that dialer must be a member of a pagegroup? At the beggining I need to setup freeswitch to stream a SIP call, so applications like VLC could listen it. Best Regards, Aleksei Zjabkin IT lahendused \ IT solutions GSM +372 58055975 ------------------------- Telegrupp AS Tartu mnt. 63, 10115 Tallinn Eesti \ Estonia