Scott, I checked the configuration of X-lite. I'm not sure if I disabled the NAT traversal stuff on the client side, but I did disable ICE, and another setting to tell the client to use the private address instead of the global address (not sure how that affects things), but when I did that I still could not hear voice mail prompts when I connected to another user's voice mail.
I've collected logs of one session and made it available here: http://www.cofc.us/merged_xml.zip It includes the registration of my soft phone with the server, and a call to 8777 to open a voice mail session. The call starts at frame 81. I don't know enough about SIP yet to know what's in the logs, but I couldn't tell that any messages weren't being delivered from the server to the client, or otherwise. Maybe you can see something there. I hope so. Gil On 5/7/09 2:07 PM, "Scott Lawrence" <[email protected]> wrote: > Make sure that you have _disabled_ any features on the phones designed > to enable NAT traversal (the server side will do it, and having both try > will mess things up). > > Take network traces - The first thing I'd do is to look at the signaling > from the server side: > > http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewe > r#Getting_SIP_Messages_to_display > > when you get the trace data, take a look at it using sipviewer > and/or post the trace with a description of your configuration > (identify components by IP address), what you were doing, and > which call in the trace you're talking about (by call-id or > frame number in the trace, preferably). > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
