[Freeswitch-users] Freeswitch Newbie Questions

[michael papet]

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 at 8000hz 1 channels 20ms
2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel
sofia/zed/mpapet at 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.
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