Our phone all Polycom phone and we use *'s transfer function rather than phone's one. We also has canreinvite=no. I believe that it is something wrong with Call Bridge between two channels(ZAP/SIP/Local). Before we didn't disable autofallthrough (default is yes), we also experienced call drop.
>I have seen this when Polycom has to communicate with none polycom >phones and a transfer is initiated to a polycom, unless the Polycom >presses Hold and then unhold, there is only one way audio, this is >without NAT involved. There might also be other cases when this >happens. My workaround is to add canreinvite=no Isaac Xiao _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
