[Freeswitch-users] Phone registration error

Michael Jerris mike at jerris.com
Mon Jul 14 20:16:47 PDT 2008 

On Jul 14, 2008, at 9:18 PM, Jair Santos wrote:

> 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?

you have no user 1001 in the 22.68.78.200 domain in the user  
directory.  If you are looking to use users from a different domain,  
you can use the force-register-domain parameter.

Mike

>
> 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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