I am having a problem with D50 phone. I configured it on the Digium Switchvox and I am using second extension that I connect to our test Asterisk server. I have one way audio from Asterisk to the phone. I do have perfect audio stream from the phone to the Asterisk server. I connected my other phone Polycom VVX500 to the same extension and the problem goes away. IS there something I need to add to Asterisk like a package to enable Digium phones ? Does anybody have any ideas. The calls I am making are extension to extension and extension to outside via SIP provider. From the network trace I see RTP packets going into the Asterisk but nothing coming back. The regular SIP (5060) packets are present both ways only the RTP is missing from Asterisk to the phone.
The other thing is that the working Polycom is configured via custom configuration from ftp and not the Switchvox system so it is possible that the configuration on the Switchvox system is causing it.
Any help would be appreciated.