[Freeswitch-users] Phone registration error

[Jair Santos]

I am trying again this message.



Hi all,
 
I've  created the following internal2.xml in the sip_profiles in order to
register a phone outside the network (NAT involved). 
I am getting Registration error 403 forbidden in the phone and "[WARNING]
sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001 at 22.68.78.200]"
on FS console.
 
 
1- Does anybody knows what am I missing?
2 -Note that the IP is hard coded ? where I have to set the Ip in order to
leave the parameters like
 
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>

Thank you
 
Jair Santos
 
 
 
 
 
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal2">
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>
 
  <aliases>
    <alias name="outbound"/>
  </aliases>
 
  <domains>
    <domain name="$${domain}" parse="true"/>
  </domains>
 
  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5060"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="manage-presence" value="false"/>
    <param name="aggressive-nat-detection" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-timeout-sec" value="1800"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="22.68.78.200"/>
    <param name="ext-sip-ip" value="22.68.78.200"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

 
 
 

J


 
 
 


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> Behalf Of faraz khan
> Sent: Monday, July 14, 2008 11:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Phone registration error
> 
> 
> 1) in vars.xml
> 
> 2) you need to make sure the domain of the user created 
> matches that of the phone thats trying to register. Multiple 
> sip 'domains' can be handled in Freeswitch.
> 
> 
> Jair Santos wrote:
> > Hi all,
> >  
> > I've  created the following internal2.xml in the 
> sip_profiles in order
> > to register a phone outside the network (NAT involved).
> > I am getting Registration error 403 forbidden in the phone and 
> > "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user 
> > [1001 at 22.68.78.200]" on FS console.
> >  
> >  
> > 1- Does anybody knows what am I missing?
> > 2 -Note that the IP is hard coded ? where I have to set the Ip in
> > order to leave the parameters like
> >  
> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >     <param name="ext-sip-ip" value="$${external_sip_ip}"/> Thank you
> >  
> > Jair Santos
> >  
> >  
> >  
> >  
> >  
> > <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
> > <profile name="internal2">
> >   <!-- This profile is only for outbound registrations to 
> providers -->
> >   <gateways>
> >     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
> >   </gateways>
> >  
> >   <aliases>
> >     <alias name="outbound"/>
> >   </aliases>
> >  
> >   <domains>
> >     <domain name="$${domain}" parse="true"/>
> >   </domains>
> >  
> >   <settings>
> >     <param name="debug" value="0"/>
> >     <param name="sip-trace" value="no"/>
> >     <param name="rfc2833-pt" value="101"/>
> >     <param name="sip-port" value="5060"/>
> >     <param name="dialplan" value="XML"/>
> >     <param name="context" value="public"/>
> >     <param name="dtmf-duration" value="100"/>
> >     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
> >     <param name="hold-music" value="$${hold_music}"/>
> >     <param name="use-rtp-timer" value="true"/>
> >     <param name="rtp-timer-name" value="soft"/>
> >     <param name="manage-presence" value="false"/>
> >     <param name="aggressive-nat-detection" value="true"/>
> >     <param name="inbound-codec-negotiation" value="generous"/>
> >     <param name="nonce-ttl" value="60"/>
> >     <param name="auth-calls" value="false"/>
> >     <param name="rtp-timeout-sec" value="1800"/>
> >     <param name="rtp-ip" value="$${local_ip_v4}"/>
> >     <param name="sip-ip" value="$${local_ip_v4}"/>
> >     <param name="ext-rtp-ip" value="22.68.78.200"/>
> >     <param name="ext-sip-ip" value="22.68.78.200"/>
> >     <param name="rtp-timeout-sec" value="300"/>
> >     <param name="rtp-hold-timeout-sec" value="1800"/>
> >   </settings>
> > </profile>
> >  
> >  
> >  
> > 
> ----------------------------------------------------------------------
> > --
> >
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> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >   
> 
> 
> 
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