in conf/directory/default.xml : <include> <!--the domain or ip (the right hand side of the @ in the addr-->
<domain name="sip.emergen.biz"> <<<<< <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"/> </params>
<variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> ~ Michael Jerris wrote:
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@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="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="22.68.78.200"/> <param name="ext-sip-ip" value="22.68.78.200"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>
J
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of faraz khan Sent: Monday, July 14, 2008 11:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phone registration error
1) in vars.xml
2) you need to make sure the domain of the user created matches that of the phone thats trying to register. Multiple sip 'domains' can be handled in Freeswitch.
Jair Santos wrote:
Hi all,
I've created the following internal2.xml in the sip_profiles in order to register a phone outside the network (NAT involved). I am getting Registration error 403 forbidden in the phone and "[WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [1001@22.68.78.200]" on FS console.
1- Does anybody knows what am I missing? 2 -Note that the IP is hard coded ? where I have to set the Ip in order to leave the parameters like
<param name="ext-rtp-ip" value="$${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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-- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz