[Freeswitch-users] Setting up port audio for incoming/outgoing calls

[Jason White]

On the wiki, an example of a port audio configuration is given that involves
creating a Sip gateway on localhost. As I couldn't get this to work
(apparently due to the external profile's detection of NAT), I thought I would
try an alternative approach. I am modifying the default dial plan here. At
some point I'll probably just rewrite it anyway.

I have created a user in the directory for extension 1020. For outbound calls,
in default.xml, I have the following:

    <extension name="portaudio" continue="true">
      <condition field="source" expression="mod_portaudio">
        <action application="set_user" data="1020@$${domain}"/>
        <action application="set" data="effective_caller_id_number=1020"/>
      </condition>
    </extension>

The log shows that the set_user is executed, as is the set
effective_caller_id_number (the latter shouldn't be necessary, unless I'm
misunderstanding).

However, running show channels after making a call from the portaudio device
still shows the user name and caller id as
FreeSWITCH,0000000000

Also, when I try to call a local extension from the audio device, I get the
following in the logs, and the call is terminated. I've checked the code, and
clearly the failure to open the file is the cause of the termination. The Sip
phone on the extension rings once and then it receives the cancellation from
FreeSWITCH.

2008-12-25 10:29:59 [DEBUG] switch_ivr_originate.c:1313 switch_ivr_originate()
P
lay Ringback File [local_stream://moh]
2008-12-25 10:29:59 [ERR] mod_local_stream.c:308 local_stream_file_open()
Unknow
n source moh
2008-12-25 10:29:59 [ERR] switch_ivr_originate.c:1322 switch_ivr_originate()
Err
or Playing File
2008-12-25 10:29:59 [DEBUG] switch_core_codec.c:122
switch_core_session_set_read
_codec() Restore original codec.
2008-12-25 10:29:59 [NOTICE] switch_ivr_originate.c:1560
switch_ivr_originate()
Hangup sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us
[CS_CONSUME_MEDIA]
 [NO_ANSWER]
2008-12-25 10:29:59 [DEBUG] switch_channel.c:1494
switch_channel_perform_hangup(
) Send signal sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us [KILL]

Any hints would be welcome.  There is no urgency, of course, as I'm doing this
for fun and out of interest.

Happy holidays to all on the FreeSWITCH list.