[Asterisk-Users] 1.2.9.1 SIP/Local/Queue behaviours weird

[Isaac Xiao]

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