[Freeswitch-users] Phone registration error
Jair Santos
jairds at shaw.ca
Mon Jul 14 11:17:15 PDT 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 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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>
>
>
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