Thanks.  It seems that outbound call to PSTN is not supported by iptel/ipkall.  Inbound is working though.  So I'm now trying to configure the outbound call through google voice but it's not connecting. The source code was downloaded yesterday via git:
git clone git://git.freeswitch.org/freeswitch.git

I pulled some other information below but it didn't help me to find the cause.  Any suggestions for next steps?  thanks

Sam

*****************************************************************************

--DingaLing status--
login    |    connected
user@gmail.com/talk    |    UNCONNECTED

*****************************************************************************

conf/jingle_profiles/client.xml

<include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
  <profile type="client">
    <param name="name" value="$${xmpp_client_profile}"/>
    <param name="login" value="user@gmail.com/talk"/>
    <param name="password" value="password"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
    <param name="server" value="talk.google.com"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="888"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>

*****************************************************************************

Enabling debug on dingaling.conf.xml:

2010-10-04 00:59:22.243782 [NOTICE] libdingaling.c:1368 SEND:
-------------------------------------------------------------------------------
<stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" to="gmail.com" version="1.0"></stream:stream>

2010-10-04 00:59:22.277462 [INFO] libdingaling.c:1366 RECV:
-------------------------------------------------------------------------------
<stream:stream from="gmail.com" id="9EAEB6231C3BDA1A" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client">
  <stream:features>
    <starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls">
      <required></required>
    </starttls>
    <mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl">
      <mechanism>X-GOOGLE-TOKEN</mechanism>
    </mechanisms>
  </stream:features>
</stream:stream>



On Fri, Oct 1, 2010 at 9:11 AM, Brian West <brian@freeswitch.org> wrote:
The far side rejected the call turn on the sip trace and watch it... It'll be rather clear what is going on if you run "sofia profile external siptrace on"

/b

On Oct 1, 2010, at 1:12 AM, Tech Man wrote:

> I'm trying to configure an iptel SIP account but everytime I call an external number, it gets rejected.  I'm running FreeSwitch 1.0.6 on a Mac OS X 10.5.8.  Any help would be greatly appreciated.  Thanks
>
> Sam


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