Re: [Freeswitch-users] Phone registration error

14 Jul 2008

      Thank you,
...
...
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.
But how can I do that? Do I have to edit the user information in the
directory. What parameter ?

And in vars.html 

 <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

Just replace auto by the IP ?

Jair Santos
...
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org 
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On 
Behalf Of faraz khan
Sent: Monday, July 14, 2008 11:08 AM
To: freeswitch-users@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@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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