<HTML><BODY STYLE="font:10pt verdana; border:none;"><DIV> <DIV>Hello</DIV> <DIV> </DIV> <DIV> I have asterisk connected to a cisco gateway and a broadsoft softswitch. My </DIV> <DIV>SIP phones are custom software on a PC. I make a call via the gateway into </DIV> <DIV>asterisk and a 2-way talk path is setup from the gateway to the sip PC. Next </DIV> <DIV>the PC (PC1) uses a SIP refer to transfer the call to the other PC (PC2)gateway</DIV> <DIV>hears ringing and then when PC2 answers, only a 1-way talk path remains </DIV> <DIV>between PC2 and the gateway. The following behavior is noted in asterisk </DIV> <DIV>starting from the call transfer:</DIV> <DIV> </DIV> <DIV> </DIV> <DIV>1) Dial the number from PC2 and click mute transfer (initiates the transfer)</DIV> <DIV> </DIV> <DIV>2) An invite is reeived at asterisk for Call Id (1) which causes 1 side of audio rtp </DIV> <DIV>stream to move to medea server in order to hear ringing.</DIV> <DIV> </DIV> <DIV>3) Asterisk in turn sends an invite to Call Id (2) in order to move other side of </DIV> <DIV>the rtp stream to media server. PC2 is hearing ringing.<BR><BR>4) PC2 answers and an invite is sent to asterisk with content-len = 0 (no rtp </DIV> <DIV>stream specified).</DIV> <DIV> </DIV> <DIV>5) Asterisk receives an ack to above invite and the rtp steam port is specified. A 1-way talk path is now setup.</DIV> <DIV> </DIV> <DIV>No other messages come in or out untill hangup occurs Seems like after the ack</DIV> <DIV>asterisk should move the other side in order to setup the other side of the talk </DIV> <DIV>path.</DIV> <DIV> </DIV> <DIV>Is there a bug in asterisk or is there a different way the transfer should be </DIV> <DIV>done.</DIV> <DIV> </DIV> <DIV>Thanks In advance For any help</DIV> <DIV>Arnie</DIV><BR><BR></DIV></BODY></HTML>