...
Sofia Profiles
Create two sip profiles, changing the SIP domain from sip.corp.ezuce.com to the SIP domain of your sipXecs systeminternal and external:
Code Block |
---|
language | html/xml |
---|
title | /etc/freeswitch/sip_profiles/internal.xml |
---|
|
<profile name="sip.corp.ezuce.comto-sipx">
<aliases>
<!--
<alias name="outbound"/>
<alias name="nat"/>
-->
</aliases>
<domains>
<domain name="all" alias="false" 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="$${internal_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="private"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="G722,PCMU@20i,PCMA@20i,speex,L16"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<param name="inbound-late-negotiation" value="true"/>
<!--<param name="enable-100rel" value="true"/>-->
<!-- This could be set to "passive" -->
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
-->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<!--<param name="presence-hosts" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
<!--<param name="force-register-domain" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
<!--all inbound reg will stored in the db using this domain -->
<!--<param name="force-register-db-domain" value="$org.sipfoundry.sipxconfig.domain.Domain@1"/>-->
<!-- ************************************************* -->
<!--<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="accept-blind-auth" value="true"/>
<!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="auto-nat$${local_ip_v4}"/>
<param name="ext-sip-ip" value="auto-nat$${local_ip_v4}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!-- <param name="enable-3pcc" value="true"/> -->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${external_ssl_enable}"/>
<!-- additional bind parameters for TLS -->
<param name="tls-bind-params" value="transport=tls"/>
<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
<param name="tls-sip-port" value="$${external_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${external_ssl_dir}"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
</settings>
<gateways>
<gateway name="sip.corp.ezuce.com">
<param name="proxy" value="sip.corp.ezuce.com"/>
<param name="realm" value="sip.corp.ezuce.com"/>
<param name="username" value="~~id~media"/>
<param name="password" value="hHcmlhKtwC"/>
-->
<param name="registertls-version" value="false$${sip_tls_version}"/>
</gateway>
</gateways>settings>
</profile> |
Code Block |
---|
language | html/xml |
---|
title | /etc/freeswitch/sip_profiles/external.xml |
---|
|
<profile name="external">
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<aliases>
<!--
<alias name="outbound"/>
<alias name="nat"/>
-->
</aliases>
<domains>
<domain name="all" alias="false" parse="true"/>
</domains>
<settings>
<param name="debug" value="1"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown-on-fail" value="true"/> -->
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<param name="rfc2833-pt" value="101"/>
<!-- RFC 5626 : Send reg-id and sip.instance -->
<!--<param name="enable-rfc-5626" value="true"/> -->
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<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="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<!--<param name="enable-100rel" value="true"/>-->
<!--<param name="disable-srv503" value="true"/>-->
<!-- This could be set to "passive" -->
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
-->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<!--<param name="presence-hosts" value="$${domain}"/>-->
<!--<param name="force-register-domain" value="$${domain}"/>-->
<!--all inbound reg will stored in the db using this domain -->
<!--<param name="force-register-db-domain" value="$${domain}"/>-->
<!-- ************************************************* -->
<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="inbound-late-negotiation" value="true"/>
<param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
<!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-sip-ip" value="auto-nat"/>
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!--<param name="enable-3pcc" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${external_ssl_enable}"/>
<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
<param name="tls-only" value="false"/>
<!-- additional bind parameters for TLS -->
<param name="tls-bind-params" value="transport=tls"/>
<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
<param name="tls-sip-port" value="$${external_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${external_ssl_dir}"/>
<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
<param name="tls-passphrase" value=""/>
<!-- Verify the date on TLS certificates -->
<param name="tls-verify-date" value="true"/>
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="none"/>
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name="tls-verify-depth" value="2"/>
<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
<param name="tls-verify-in-subjects" value=""/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
</settings>
</profile> |
...
Code Block |
---|
language | html/xml |
---|
title | /etc/freeswitch/dialplan/default.xml |
---|
|
<include>
<context name="private">
<extension name="going-out">
<condition field="destination_number" expression="^(\d{7,20})$">
<action application="set" data="transfer_ringback=$${us-ring}"/>
<!--Set this for a generic Caller ID -->
<action application="set" data="effective_caller_id_number=5553211234"/>
<action application="bridge" data="sofia/gateway/voip.ms/$1"/>
<action application="answer"/>
<anti-action application="set" data="transfer_ringback=$${us-ring}"/>
<anti-action application="bridge" data="sofia/gateway/sipto-sipx/${destination_number}@sip.corp.ezuce.com/${destination_number}"/>
<anti-action application="answer"/>
</condition>
</extension>
</context>
</include> |
...
Code Block |
---|
language | html/xml |
---|
title | /etc/freeswitch/dialplan/public.xml |
---|
|
<include>
<context name="public">
<extension name="going-in">
<condition>
<action application="set" data="transfer_ringback=$${us-ring}"/>
<action application="bridge" data="{ignore_early_media=true}sofia/gateway/sipto-sipx/${destination_number}@sip.corp.ezuce.com/${destination_number}"/>
<action application="answer"/>
</condition>
</extension>
</context>
</include>
|
...