Okay, so I've figured out one or two things but have stumbled across a minor issue. I've got the following as my test plan: <!--#include "default_context.xml"--> <context name="default"> <extension name="playsound1" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="caller_id_name" expression="mpapet"> <action application="answer"/> <action application="playback" data="/var/sounds/congrats.wav"/> </condition> </extension> 2007-02-02 13:00:36 [NOTICE] switch_core.c:2802 switch_core_standard_on_execute() Execute playback(/var/sounds/congrats.wav) 2007-02-02 13:00:36 [DEBUG] mod_sndfile.c:138 sndfile_file_open() Opening File [/var/sounds/congrats.wav] 8000hz 2007-02-02 13:00:36 [DEBUG] switch_ivr.c:1125 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel sofia/zed/mpapet@voip.katigo.mooo.com entering state [completed] It answers and starts playing the file according to the console but I hear no sound. Do I need some kind of other statement in there? Or, is it a NAT issue. Or? What does this error message mean? [DEBUG] mod_sofia.c:4585 event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a FYI: If I remove freeswitch_combined.xml, the file default_context.xml does not get read/included from freeswitch.xml. Something like "cannot find default context" kept appearing in the console despite the presences of default_context.xml. My quick and dirty solution was to put the default context inside freeswitch.xml. I verified this in the freeswitch console with the error message. After the change, call works great. As an FYI for future users, I'm running openser and freeswitch on the same machine using virtual ip addresses. You need to modify sofia.conf.xml<param name="sip-ip" value=" your.freeswitch.virtual-ip.address"/> to get it to register freeswitch to your sip server.