Hi Elliot, IIRC, the scenario where the Polycom is behind NAT and the FS is on a public IP is one where Polycom fails because they are too stubborn to add rport to their SIP handling. (We've been barking at them for over 5 years on this issue.) There is a workaround in FS but it breaks non-Polycom phones that are on the same profile, so you'll need to have a separate profile for your Polycoms if you are in a mixed environment. In any case, for the sofia profile that has the Polycoms you should check out these two parameters: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#NDLB-force-rport http://wiki.freeswitch.org/wiki/NAT_Traversal#NDLB-connectile-dysfunction You might also want to hop into #freeswitch on irc.freenode.net and talk to some of our resident Polycom veterans. -MC On Wed, Nov 16, 2011 at 9:14 PM, Elliott Vogel <elliott@zoogmedia.com>wrote:
Hello, I’m trying to setup freeswitch and I’m having a problem with a polycom phone registering. The phone is behind nat but the freeswitch is on a public network can anyone tell me if I don’t have something configured correctly or where should look for the problem?****
** **
<include>****
<domain name="zoogmedia.com">****
<params>****
<param name="dial-string" value=" {sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>****
</params>****
** **
<variables>****
<variable name="record_stereo" value="true"/>****
<variable name="default_gateway" value="$${default_provider}"/>****
<variable name="default_areacode" value="312"/>****
<variable name="transfer_fallback_extension" value="operator"/>****
</variables>****
** **
<groups>****
<group name="default">****
<users>****
<user id="14149823263">****
<params>****
<param name="password" value="123456"/>****
</params>****
<variables>****
<variable name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>****
<variable name="toll_allow" value=" domestic,international,local"/>****
<variable name="accountcode" value="1000"/>****
<variable name="user_context" value="default"/>****
<variable name="outbound_caller_id_name" value="Sales" />*** *
<variable name="outbound_caller_id_number" value=" +14149823263" />****
</variables>****
</user>****
</users>****
</group>****
</groups>****
** **
</domain>****
</include>****
** **
** **
** **
<configuration name="sofia.conf" description="sofia Endpoint">****
** **
<global_settings>****
<param name="log-level" value="0"/>****
<!-- <param name="auto-restart" value="false"/> -->****
<param name="debug-presence" value="0"/>****
<!-- <param name="capture-server" value="udp:homer.domain.com:5060"/> -->****
</global_settings>****
** **
<profiles>****
** **
<profile name="endpoints">****
<gateways>****
<gateway name="TNG">****
<param name="username" value="user"/>****
<param name="password" value="password"/>****
<param name="proxy" value="69.25.128.195:5060"/>****
<param name="from-domain" value="$${local_ip_v4}:5060"/>****
<param name="dtmf-type" value="rfc2833"/>****
<param name="extension-in-contact" value="true"/>****
<param name="caller-id-in-from" value="true"/>****
<param name="register" value="false"/>****
</gateway>****
</gateways>****
<aliases>****
** **
</aliases>****
<domains>****
<domain name="all" alias="false" parse="true"/>****
</domains>****
<settings>****
<param name="debug" value="0"/>****
<param name="sip-trace" value="no"/>****
<param name="sip-capture" value="no"/>****
<param name="rtp-ip" value="$${external_rtp_ip}"/>****
<param name="sip-ip" value="$${external_sip_ip}"/>****
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>****
<param name="ext-sip-ip" value="$${external_sip_ip}"/>****
<param name="sip-port" value="5070"/>****
<param name="tls" value="$${external_ssl_enable}"/>****
<param name="tls-bind-params" value="transport=tls"/>****
<param name="tls-sip-port" value="$${external_tls_port}"/>****
<param name="tls-cert-dir" value="$${external_ssl_dir}"/>****
<param name="tls-version" value="$${sip_tls_version}"/>****
** **
<param name="rfc2833-pt" value="101"/>****
<param name="dialplan" value="XML"/>****
<param name="context" value="default"/>****
<param name="dtmf-duration" value="2000"/>****
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> ****
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs} "/>****
<param name="hold-music" value="$${hold_music}"/>****
<param name="rtp-timer-name" value="soft"/>****
<param name="local-network-acl" value="localnet.auto"/>****
<param name="manage-presence" value="false"/>****
<param name="inbound-codec-negotiation" value="greedy"/>****
<param name="nonce-ttl" value="60"/>****
<param name="auth-calls" value="false"/>****
<param name="rtp-timeout-sec" value="300"/>****
<param name="rtp-hold-timeout-sec" value="1800"/>****
<param name="rtp-rewrite-timestamps" value="true"/>****
** **
<param name="track-calls" value="true"/>****
</settings>****
</profile>****
** **
</profiles>****
** **
</configuration>****
** **
<?xml version="1.0" encoding="utf-8"?>****
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->****
<include>****
<context name="default">****
<extension name="unloop">****
<condition field="${unroll_loops}" expression="^true$"/>****
<condition field="${sip_looped_call}" expression="^true$">****
<action application="deflect" data="${destination_number}"/>****
</condition>****
</extension>****
<extension name="check_auth" continue="true">****
<condition field="${sip_authorized}" expression="^true$" break=" never">****
<anti-action application="respond" data="407"/>****
</condition>****
</extension>****
<extension name="local">****
<condition field="${toll_allow}" expression="local" />****
<condition field="destination_number" expression="^([0-9]{7})$">****
<action application="set" data=" effective_caller_id_name=${outbound_caller_id_name}" />****
<action application="set" data=" effective_caller_id_number=${outbound_caller_id_number}" />****
<action application="bridge" data=" sofia/gateway/TNG/+1${default_areacode}$1"/>****
</condition>****
</extension>****
<extension name="domestic">****
<condition field="${toll_allow}" expression="domestic" />****
<condition field="destination_number" expression="^(1{0,1}\d{10})$"> ****
<action application="set" data=" effective_caller_id_name=${outbound_caller_id_name}" />****
<action application="set" data=" effective_caller_id_number=${outbound_caller_id_number}" />****
<action application="bridge" data="sofia/gateway/TNG/+$1"/>****
</condition>****
</extension>****
<extension name="international">****
<condition field="${toll_allow}" expression="international" />****
<condition field="destination_number" expression="^011?(\d+)$">****
<action application="set" data=" effective_caller_id_name=${outbound_caller_id_name}" />****
<action application="set" data=" effective_caller_id_number=${outbound_caller_id_number}" />****
<action application="bridge" data="sofia/gateway/TNG/+$1"/>****
</condition>****
</extension>****
</context>****
</include>****
** **
** **
** **
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org