<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3354" name=GENERATOR></HEAD>
<BODY><!-- Converted from text/plain format -->
<P><FONT size=2>Hi ,<BR><BR>I am strugling to make two phones to make calls. I
'll appreciate information about the configs that I am listing below.
<BR><BR><STRONG>1s phone</STRONG> - Is an internat phone(behind NAT). This one
was working before. It was able to make calls to PSTN and to another
extension.<BR><BR>The config is like<BR><BR>Sip server address:
192.168.1.117<BR>Sip user id : 1000<BR><BR><STRONG>2nd phone</STRONG> - It is
external the of the Network.<BR><BR><BR>The config is like :<BR>Username :
1001<BR>Domain 24.67.78.200:5060<BR><BR><BR>Both are registering in FS. I cannot
call between them.<BR><BR><STRONG>When I try to make PSTN calls using
external phone I get .<BR></STRONG><BR><BR>2008-07-15 10:45:34 [NOTICE]
switch_channel.c:533 switch_channel_set_name() New Channel
sofia/internal2/1001@24.67.78.200:5060
[58e39d4a-e78f-4c96-8f43-3e969704d9e1]<BR>2008-07-15 10:45:34 [INFO]
mod_dialplan_xml.c:222 dialplan_hunt() Processing
FreeSwitch->12508606894@public<BR>2008-07-15 10:45:34 [ERR] mod_sofia.c:1713
sofia_outgoing_channel() Invalid Gateway<BR>2008-07-15 10:45:34 [NOTICE]
mod_sofia.c:1904 sofia_outgoing_channel() Close Channel N/A
[CS_NEW]<BR>2008-07-15 10:45:34 [ERR] switch_ivr_originate.c:823
switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause:
[INVALID_NUMBER_FORMAT]<BR>2008-07-15 10:45:34 [INFO] mod_dptools.c:1797
audio_bridge_function() Originate Failed. Cause:
INVALID_NUMBER_FORMAT<BR>2008-07-15 10:45:34 [NOTICE] mod_dptools.c:1824
audio_bridge_function() Hangup sofia/internal2/1001@24.67.78.200:5060
[CS_EXECUTE] [INVALID_NUMBER_FORMAT]<BR>2008-07-15 10:45:34 [NOTICE]
switch_core_session.c:753 switch_core_session_thread() Session 17
(sofia/internal2/1001@24.67.78.200:5060) Ended<BR>2008-07-15 10:45:34 [NOTICE]
switch_core_session.c:755 switch_core_session_thread() Close Channel
sofia/internal2/1001@24.67.78.200:5060 [CS_HANGUP]<BR><BR><BR><FONT
size=3><STRONG>Please check the settings that I
have.<BR></STRONG><BR></FONT><STRONG>On
/conf/dialplan/extensions</STRONG><BR></FONT><FONT size=2>teste.xml</P>
<P><FONT face=Tahoma color=#0000ff></FONT><FONT face=Tahoma
color=#0000ff></FONT><BR><extension name="teste"><BR>
<condition field="destination_number"
expression="^(\d+)$"><BR> <!--- The %
behind the username tells FS to lookup the user in it's local sip_registration
database --><BR> <action
application="bridge" data="sofia/gateway/inphonex/$0"/><BR>
</condition><BR> </extension><BR><BR><BR><STRONG>On
conf/sip_profiles</STRONG></FONT></P>
<P><FONT size=2><STRONG>internal2.xml</STRONG></FONT></P>
<P><FONT size=2><BR><BR><!-- <A
href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files</A>
--><BR><profile name="internal2"><BR> <!-- This profile is
only for outbound registrations to providers --><BR>
<gateways><BR> <X-PRE-PROCESS cmd="include"
data="external/*.xml"/><BR> </gateways><BR><BR>
<aliases><BR> <alias name="internal2"/><BR>
</aliases><BR><BR> <domains><BR> <domain
name="$${domain}" parse="true"/><BR> </domains><BR><BR>
<settings><BR> <param name="debug"
value="0"/><BR> <param name="sip-trace"
value="no"/><BR> <param name="rfc2833-pt"
value="101"/><BR> <param name="sip-port"
value="5060"/><BR> <param name="dialplan"
value="XML"/><BR> <param name="context"
value="public"/><BR> <param name="dtmf-duration"
value="100"/><BR> <param name="codec-prefs"
value="$${outbound_codec_prefs}"/><BR> <param
name="hold-music" value="$${hold_music}"/><BR> <param
name="use-rtp-timer" value="true"/><BR> <param
name="rtp-timer-name" value="soft"/><BR> <param
name="manage-presence" value="false"/><BR> <param
name="aggressive-nat-detection" value="true"/><BR>
<param name="inbound-codec-negotiation"
value="generous"/><BR> <param name="nonce-ttl"
value="60"/><BR> <param name="auth-calls"
value="false"/><BR> <param name="rtp-timeout-sec"
value="1800"/><BR> <param name="rtp-ip"
value="$${local_ip_v4}"/><BR> <param name="sip-ip"
value="$${local_ip_v4}"/><BR> <param name="ext-rtp-ip"
value="24.67.78.200"/><BR> <param name="ext-sip-ip"
value="24.67.78.200"/><BR> <param name="rtp-timeout-sec"
value="300"/><BR> <param name="rtp-hold-timeout-sec"
value="1800"/><BR> </settings><BR></profile><BR></P></FONT>
<DIV><FONT size=2><STRONG>doublenat.xml</STRONG></FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff><!-- <A
href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files</A>
--><BR><profile name="doublenat"><BR> <!-- This profile is
only for outbound registrations to providers --><BR>
<gateways><BR> <X-PRE-PROCESS cmd="include"
data="external/*.xml"/><BR> </gateways></FONT></FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff>
<aliases><BR> <alias name="doublenat"/><BR>
</aliases></FONT></FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff>
<domains><BR> <domain name="$${domain}"
parse="true"/><BR> </domains></FONT></FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff>
<settings><BR> <param name="debug"
value="0"/><BR> <param name="force-register-domain"
value="$${domain}"/><BR> <param name="apply-nat-acl"
value="rfc1918"/><BR> <param name="sip-trace"
value="no"/><BR> <param name="rfc2833-pt"
value="101"/><BR> <param name="sip-port"
value="5090"/><BR> <param name="dialplan"
value="XML"/><BR> <param name="context"
value="public"/><BR> <param name="dtmf-duration"
value="100"/><BR> <param name="codec-prefs"
value="$${outbound_codec_prefs}"/><BR> <param
name="hold-music" value="$${hold_music}"/><BR> <param
name="use-rtp-timer" value="true"/><BR> <param
name="rtp-timer-name" value="soft"/><BR> <param
name="manage-presence" value="false"/><BR> <param
name="aggressive-nat-detection" value="true"/><BR>
<param name="inbound-codec-negotiation"
value="generous"/><BR> <param name="nonce-ttl"
value="60"/><BR> <param name="auth-calls"
value="false"/><BR> <param name="rtp-timeout-sec"
value="1800"/><BR> <param name="rtp-ip"
value="$${local_ip_v4}"/><BR> <param name="sip-ip"
value="$${local_ip_v4}"/><BR> <param name="ext-rtp-ip"
value="$${external_rtp_ip}"/><BR> <param
name="ext-sip-ip" value="$${external_sip_ip}"/><BR>
<param name="rtp-timeout-sec" value="300"/><BR>
<param name="rtp-hold-timeout-sec" value="1800"/><BR>
</settings><BR></profile><BR></FONT></FONT></DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff></FONT></FONT> </DIV>
<DIV><FONT size=2><FONT face=Tahoma color=#0000ff></FONT></FONT> </DIV>
<DIV><FONT size=2><FONT face=Tahoma><STRONG>On
/conf/directory/default<BR></STRONG></FONT></FONT></DIV>
<DIV><FONT size=2><FONT face=Tahoma>inphonex.xml</DIV></FONT>
<P><FONT face=Tahoma color=#0000ff><include><BR> <user
id="inphonex"><BR>
<gateways><BR> <gateway
name="inphonex"><BR>
<param name="username"
value="3462101"/><BR>
<param name="realm"
value="sip.varphonex.com"/><BR>
<param name="password"
value="606545"/><BR>
<param name="proxy"
value="sip.varphonex.com"/><BR>
<param name="register"
value="true"/><BR>
<param name="expire-seconds"
value="3600"/><BR>
</gateway><BR></gateways><BR><params><BR>
<param name="password" value="606545"/><BR>
</params><BR>
</user><BR></include><BR></FONT><BR><FONT face=Tahoma
color=#0000ff>Thank you </FONT></P>
<P><FONT face=Tahoma color=#0000ff>Jair santos</FONT></P></FONT></BODY></HTML>