From 1fbe4aef121120151bdf0610ae8355342b40e6b6 Mon Sep 17 00:00:00 2001
From: Keith Whyte <keith@rhizomatica.org>
Date: Wed, 2 Aug 2023 05:52:08 +0200
Subject: [PATCH] FS: Rename SIP Profiles

---
 .../sip_profiles/internal/example.xml         |  37 --
 .../freeswitch/sip_profiles/internalvpn.xml   | 412 ------------------
 .../files/etc/freeswitch/sip_profiles/lan.xml |   6 +-
 .../sip_profiles/{external.xml => vpn.xml}    |  31 +-
 modules/rhizo_base/manifests/freeswitch.pp    |  40 +-
 modules/rhizo_base/templates/GERAN.xml.erb    | 191 ++++++++
 modules/rhizo_base/templates/extrtp.xml.erb   |  86 ++++
 .../templates/monitor_freeswitch.sh.erb       |  33 +-
 .../rem_xcode.xml.erb}                        |   8 +-
 modules/rhizo_base/templates/vars.xml.erb     |   3 +-
 10 files changed, 333 insertions(+), 514 deletions(-)
 delete mode 100644 modules/rhizo_base/files/etc/freeswitch/sip_profiles/internal/example.xml
 delete mode 100644 modules/rhizo_base/files/etc/freeswitch/sip_profiles/internalvpn.xml
 rename modules/rhizo_base/files/etc/freeswitch/sip_profiles/{external.xml => vpn.xml} (84%)
 create mode 100644 modules/rhizo_base/templates/GERAN.xml.erb
 create mode 100644 modules/rhizo_base/templates/extrtp.xml.erb
 rename modules/rhizo_base/{files/etc/freeswitch/sip_profiles/rem_xcode.xml => templates/rem_xcode.xml.erb} (88%)

diff --git a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internal/example.xml b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internal/example.xml
deleted file mode 100644
index 10263b0..0000000
--- a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internal/example.xml
+++ /dev/null
@@ -1,37 +0,0 @@
-<include>
-  <!--<gateway name="asterlink.com">-->
-  <!--/// account username *required* ///-->
-  <!--<param name="username" value="cluecon"/>-->
-  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
-  <!--<param name="realm" value="asterlink.com"/>-->
-  <!--/// username to use in from: *optional* same as  username, if blank ///-->
-  <!--<param name="from-user" value="cluecon"/>-->
-  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
-  <!--<param name="from-domain" value="asterlink.com"/>-->
-  <!--/// account password *required* ///-->
-  <!--<param name="password" value="2007"/>-->
-  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
-  <!--<param name="extension" value="cluecon"/>-->
-  <!--/// proxy host: *optional* same as realm, if blank ///-->
-  <!--<param name="proxy" value="asterlink.com"/>-->
-  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
-  <!--<param name="register-proxy" value="mysbc.com"/>-->
-  <!--/// expire in seconds: *optional* 3600, if blank ///-->
-  <!--<param name="expire-seconds" value="60"/>-->
-  <!--/// do not register ///-->
-  <!--<param name="register" value="false"/>-->
-  <!-- which transport to use for register -->
-  <!--<param name="register-transport" value="udp"/>-->
-  <!--How many seconds before a retry when a failure or timeout occurs -->
-  <!--<param name="retry-seconds" value="30"/>-->
-  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
-  <!--<param name="caller-id-in-from" value="false"/>-->
-  <!--extra sip params to send in the contact-->
-  <!--<param name="contact-params" value="tport=tcp"/>-->
-  <!-- Put the extension in the contact -->
-  <!--<param name="extension-in-contact" value="true"/>-->
-  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
-  <!--<param name="ping" value="25"/>-->
-  <!--<param name="cid-type" value="rpid"/>-->
-  <!--</gateway>-->
-</include>
diff --git a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internalvpn.xml b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internalvpn.xml
deleted file mode 100644
index d62391c..0000000
--- a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/internalvpn.xml
+++ /dev/null
@@ -1,412 +0,0 @@
-<profile name="internalvpn">
-  <!--
-      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
-      In FreeSWITCH you can run multiple sip user agents on their own ip and port.
-
-      When you hear someone say "sofia profile" this is what they are talking about.
-  -->
-
-  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
-  <!--aliases are other names that will work as a valid profile name for this profile-->
-  <aliases>
-    <!--
-        <alias name="default"/>
-    -->
-  </aliases>
-  <!-- Outbound Registrations -->
-
-  <domains>
-    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
-    <!--<domain name="$${domain}" parse="true"/>-->
-    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
-    <!--<domain name="all" alias="true" parse="true"/>-->
-    <domain name="all" alias="true" parse="false"/>
-  </domains>
-
-  <settings>
-
-
-    <!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
-    <!-- <param name="rtp-digit-delay" value="40"/>-->
-
-    <!--
-        When calls are in no media this will bring them back to media
-        when you press the hold button.
-    -->
-    <!--<param name="media-option" value="resume-media-on-hold"/> -->
-    <!--
-        This will allow a call after an attended transfer go back to
-        bypass media after an attended transfer.
-    -->
-    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
-    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
-    <param name="debug" value="0"/>
-    <!-- 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"/>
-
-    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
-    <!-- <param name="presence-proto-lookup" value="true"/> -->
-
-
-    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
-    <param name="liberal-dtmf" value="true"/>
-
-
-    <!--
-        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
-        responding. These options allow you to enable and control a watchdog
-        on the Sofia SIP stack so that if it stops responding for the
-        specified number of milliseconds, it will cause FreeSWITCH to crash
-        immediately. This is useful if you run in an HA environment and
-        need to ensure automated recovery from such a condition. Note that if
-        your server is idle a lot, the watchdog may fire due to not receiving
-        any SIP messages. Thus, if you expect your system to be idle, you
-        should leave the watchdog disabled. It can be toggled on and off
-        through the FreeSWITCH CLI either on an individual profile basis or
-        globally for all profiles. So, if you run in an HA environment with a
-        master and slave, you should use the CLI to make sure the watchdog is
-        only enabled on the master.
-        If such crash occurs, FreeSWITCH will dump core if allowed. The
-        stacktrace will include function watchdog_triggered_abort().
-    -->
-    <param name="watchdog-enabled" value="no"/>
-    <param name="watchdog-step-timeout" value="30000"/>
-    <param name="watchdog-event-timeout" value="30000"/>
-
-    <param name="log-auth-failures" value="false"/>
-    <param name="forward-unsolicited-mwi-notify" value="false"/>
-
-    <param name="context" value="public"/>
-    <param name="rfc2833-pt" value="101"/>
-    <!-- port to bind to for sip traffic -->
-    <param name="sip-port" value="5040"/>
-    <param name="dialplan" value="XML"/>
-    <param name="dtmf-duration" value="2000"/>
-    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
-    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
-    <param name="rtp-timer-name" value="soft"/>
-    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
-    <param name="rtp-ip" value="$${local_ip_v4}"/>
-    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
-    <param name="sip-ip" value="$${local_ip_v4}"/>
-    <!--<param name="hold-music" value="$${hold_music}"/>-->
-    <param name="apply-nat-acl" value="nat.auto"/>
-
-
-    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
-    <!-- <param name="cid-in-1xx" value="false"/> -->
-
-    <!-- extended info parsing -->
-    <!-- <param name="extended-info-parsing" value="true"/> -->
-
-    <param name="aggressive-nat-detection" value="false"/>
-    <!--
-        There are known issues (asserts and segfaults) when 100rel is enabled.
-        It is not recommended to enable 100rel at this time.
-    -->
-    <!--<param name="enable-100rel" value="true"/>-->
-
-    <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
-    <!-- RFC3263 Section 4.3 -->
-    <!--<param name="disable-srv503" value="true"/>-->
-
-    <!-- Enable Compact SIP headers. -->
-    <!--<param name="enable-compact-headers" value="true"/>-->
-    <!--
-        enable/disable session timers
-    -->
-    <!--<param name="enable-timer" value="false"/>-->
-    <!--<param name="minimum-session-expires" value="120"/>-->
-    <param name="apply-inbound-acl" value="domains"/>
-    <!--
-        This defines your local network, by default we detect your local network
-        and create this localnet.auto ACL for this.
-    -->
-    <param name="local-network-acl" value="localnet.auto"/>
-    <param name="apply-register-acl" value="domains"/>
-    <param name="apply-candidate-acl" value="domains"/>
-    <!--<param name="dtmf-type" value="info"/>-->
-
-
-    <!-- 'true' means every time 'first-only' means on the first register -->
-    <!--<param name="send-message-query-on-register" value="true"/>-->
-
-    <!-- 'true' means every time 'first-only' means on the first register -->
-    <!--<param name="send-presence-on-register" value="first-only"/> -->
-
-
-    <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
-    <!-- Remote-Party-ID header -->
-    <!--<param name="caller-id-type" value="rpid"/>-->
-
-    <!-- P-*-Identity family of headers -->
-    <!--<param name="caller-id-type" value="pid"/>-->
-
-    <!-- neither one -->
-    <!--<param name="caller-id-type" value="none"/>-->
-
-
-
-    <param name="record-path" value="$${recordings_dir}"/>
-    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-    <!--enable to use presence -->
-    <param name="manage-presence" value="false"/>
-    <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
-    <!--<param name="presence-probe-on-register" value="true"/>-->
-    <!--<param name="manage-shared-appearance" value="true"/>-->
-    <!-- used to share presence info across sofia profiles -->
-    <!-- Name of the db to use for this profile -->
-    <!--<param name="dbname" value="share_presence"/>-->
-    <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
-    <param name="presence-privacy" value="$${presence_privacy}"/>
-    <!-- ************************************************* -->
-
-    <!-- This setting is for AAL2 bitpacking on G726 -->
-    <!-- <param name="bitpacking" value="aal2"/> -->
-    <!--max number of open dialogs in proceeding -->
-    <!--<param name="max-proceeding" value="1000"/>-->
-    <!--session timers for all call to expire after the specified seconds -->
-    <!--<param name="session-timeout" value="1800"/>-->
-    <!-- Can be 'true' or 'contact' -->
-    <!--<param name="multiple-registrations" value="contact"/>-->
-    <!--set to 'greedy' if you want your codec list to take precedence -->
-    <param name="inbound-codec-negotiation" value="greedy"/>
-    <!-- if you want to send any special bind params of your own -->
-    <!--<param name="bind-params" value="transport=udp"/>-->
-    <!--<param name="unregister-on-options-fail" value="true"/>-->
-    <!-- Send an OPTIONS packet to all registered endpoints -->
-    <!--<param name="all-reg-options-ping" value="true"/>-->
-    <!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
-    <!--<param name="nat-options-ping" value="true"/>-->
-
-    <!-- TLS: disabled by default, set to "true" to enable -->
-    <param name="tls" value="$${internal_ssl_enable}"/>
-    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
-    <param name="tls-only" value="false"/>
-    <!-- additional bind parameters for TLS -->
-    <param name="tls-bind-params" value="transport=tls"/>
-    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
-    <param name="tls-sip-port" value="$${internal_tls_port}"/>
-    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
-    <!--<param name="tls-cert-dir" value=""/>-->
-    <!-- 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}"/>
-
-    <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
-         (reduces delay on latent connections default true, must be disabled explicitly)-->
-    <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
-
-    <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
-    <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
-    <!--<param name="pass-rfc2833" value="true"/>-->
-    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
-    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
-
-    <!-- Or, if you have PGSQL support, you can use that -->
-    <!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />-->
-
-    <!--Uncomment to set all inbound calls to no media mode-->
-    <!--<param name="inbound-bypass-media" value="true"/>-->
-
-    <!--Uncomment to set all inbound calls to proxy media mode-->
-    <!--<param name="inbound-proxy-media" value="true"/>-->
-
-    <!-- Let calls hit the dialplan before selecting codec for the a-leg -->
-    <param name="inbound-late-negotiation" value="false"/>
-
-    <!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
-    <param name="inbound-zrtp-passthru" value="true"/>
-
-    <!-- this lets anything register -->
-    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-    <!-- <param name="accept-blind-reg" value="true"/> -->
-
-    <!-- accept any authentication without actually checking (not a good feature for most people) -->
-    <!-- <param name="accept-blind-auth" value="true"/> -->
-
-    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
-    <!-- <param name="suppress-cng" value="true"/> -->
-
-    <!--TTL for nonce in sip auth-->
-    <param name="nonce-ttl" value="60"/>
-    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
-        that the originator is using-->
-    <!--<param name="disable-transcoding" value="true"/>-->
-    <!-- Handle 302 Redirect in the dialplan -->
-    <!--<param name="manual-redirect" value="true"/> -->
-    <!-- Disable Transfer -->
-    <!--<param name="disable-transfer" value="true"/> -->
-    <!-- Disable Register -->
-    <!--<param name="disable-register" value="true"/> -->
-    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
-    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
-    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
-    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
-    <param name="auth-calls" value="$${internal_auth_calls}"/>
-    <!-- Force the user and auth-user to match. -->
-    <param name="inbound-reg-force-matching-username" value="true"/>
-    <!-- on authed calls, authenticate *all* the packets not just invite -->
-    <param name="auth-all-packets" value="false"/>
-
-    <!-- external_sip_ip
-         Used as the public IP address for SDP.
-         Can be an one of:
-         ip address            - "12.34.56.78"
-         a stun server lookup  - "stun:stun.server.com"
-         a DNS name            - "host:host.server.com"
-         auto                  - Use guessed ip.
-         auto-nat              - Use ip learned from NAT-PMP or UPNP
-    -->
-    <param name="ext-rtp-ip" value="$${local_ip_v4}"/>
-    <param name="ext-sip-ip" value="$${local_ip_v4}"/>
-
-    <!-- rtp inactivity timeout -->
-    <param name="rtp-timeout-sec" value="300"/>
-    <param name="rtp-hold-timeout-sec" value="1800"/>
-    <!-- VAD choose one (out is a good choice); -->
-    <!-- <param name="vad" value="in"/> -->
-    <!-- <param name="vad" value="out"/> -->
-    <!-- <param name="vad" value="both"/> -->
-    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-    <!--
-        These are enabled to make the default config work better out of the box.
-        If you need more than ONE domain you'll need to not use these options.
-
-    -->
-    <!--all inbound reg will look in this domain for the users -->
-    <param name="force-register-domain" value="$${domain}"/>
-    <!--force the domain in subscriptions to this value -->
-    <param name="force-subscription-domain" value="$${domain}"/>
-    <!--all inbound reg will stored in the db using this domain -->
-    <param name="force-register-db-domain" value="$${domain}"/>
-
-
-    <!-- uncomment for sip over websocket support -->
-    <!--<param name="ws-binding"  value=":5066"/>-->
-
-    <!-- uncomment for sip over secure websocket support -->
-    <!-- You need wss.pem in /usr/local/freeswitch/certs for wss -->
-    <!--<param name="wss-binding" value=":7443"/>-->
-    
-
-    <!--<param name="delete-subs-on-register" value="false"/>-->
-
-    <!-- launch a new thread to process each new inbound register when using heavier backends -->
-    <!-- <param name="inbound-reg-in-new-thread" value="true"/> -->
-
-    <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
-    <param name="rtcp-audio-interval-msec" value="5000"/>
-    <!--<param name="rtcp-video-interval-msec" value="5000"/>-->
-
-    <!--force suscription expires to a lower value than requested-->
-    <!--<param name="force-subscription-expires" value="60"/>-->
-
-    <!-- add a random deviation to the expires value of the 202 Accepted -->
-    <!--<param name="sip-subscription-max-deviation" value="120"/>-->
-
-    <!-- disable register and transfer which may be undesirable in a public switch -->
-    <!--<param name="disable-transfer" value="true"/>-->
-    <!--<param name="disable-register" value="true"/>-->
-
-    <!--
-        enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
-        right away, proxy waits until the call has been answered then sends accepts
-    -->
-    <!--<param name="enable-3pcc" value="true"/>-->
-
-    <!-- use at your own risk or if you know what this does.-->
-    <!--<param name="NDLB-force-rport" value="true"/>-->
-    <!--
-        Choose the realm challenge key. Default is auto_to if not set.
-
-        auto_from  - uses the from field as the value for the sip realm.
-        auto_to    - uses the to field as the value for the sip realm.
-        <anyvalue> - you can input any value to use for the sip realm.
-
-        If you want URL dialing to work you'll want to set this to auto_from.
-
-        If you use any other value besides auto_to or auto_from you'll
-        loose the ability to do multiple domains.
-
-        Note: comment out to restore the behavior before 2008-09-29
-    -->
-    <param name="challenge-realm" value="auto_from"/>
-    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
-    <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
-    <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
-    <!-- on outbound calls set the callid to match the uuid of the session -->
-    <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
-    <!-- set to false disable this feature -->
-    <!--<param name="rtp-autofix-timing" value="false"/>-->
-
-    <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
-    <!--<param name="pass-callee-id" value="false"/>-->
-
-    <!-- clear clears them all or supply the name to add or the name
-         prefixed with ~ to remove valid values:
-
-           clear
-           CISCO_SKIP_MARK_BIT_2833
-           SONUS_SEND_INVALID_TIMESTAMP_2833
-
-    -->
-    <!--<param name="auto-rtp-bugs" data="clear"/>-->
-
-    <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
-    <!--<param name="disable-srv" value="false" />-->
-    <!--<param name="disable-naptr" value="false" />-->
-
-    <!-- The following can be used to fine-tune timers within sofia's transport layer
-         Those settings are for advanced users and can safely be left as-is -->
-
-    <!-- Initial retransmission interval (in milliseconds).
-         Set the T1 retransmission interval used by the SIP transaction engine.
-         The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G.   -->
-    <!-- <param name="timer-T1" value="500" /> -->
-
-    <!--  Transaction timeout (defaults to T1 * 64).
-         Set the T1x64 timeout value used by the SIP transaction engine.
-         The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
-         The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
-    <!-- <param name="timer-T1X64" value="32000" /> -->
-
-
-    <!-- Maximum retransmission interval (in milliseconds).
-         Set the maximum retransmission interval used by the SIP transaction engine.
-         The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
-         Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
-         until the timer B fires.  -->
-    <!-- <param name="timer-T2" value="4000" /> -->
-
-    <!--
-        Transaction lifetime (in milliseconds).
-        Set the lifetime for completed transactions used by the SIP transaction engine.
-        A completed transaction is kept around for the duration of T4 in order to catch late responses.
-        The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
-    <!-- <param name="timer-T4" value="4000" /> -->
-
-    <!-- Turn on a jitterbuffer for every call -->
-    <!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
-
-
-    <!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
-         Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
-         It's probably not what you want so stick with the default unless you really need to change this.
-    -->
-    <!--<param name="renegotiate-codec-on-hold" value="true"/>-->
-
-  </settings>
-</profile>
diff --git a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/lan.xml b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/lan.xml
index 5b1a347..4713169 100644
--- a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/lan.xml
+++ b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/lan.xml
@@ -19,7 +19,7 @@
 
     <param name="context" value="public"/>
     <param name="rfc2833-pt" value="101"/>
-    <param name="sip-port" value="5091"/>
+    <param name="sip-port" value="5060"/>
     <param name="dialplan" value="XML"/>
     <param name="dtmf-duration" value="2000"/>
     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
@@ -45,8 +45,8 @@
     <param name="auth-calls" value="$${internal_auth_calls}"/>
     <param name="inbound-reg-force-matching-username" value="true"/>
     <param name="auth-all-packets" value="false"/>
-    <param name="ext-rtp-ip" value="$${external_ip}"/>
-    <param name="ext-sip-ip" value="$${external_ip}"/>
+    <param name="ext-rtp-ip" value="$${lan_ip}"/>
+    <param name="ext-sip-ip" value="$${lan_ip}"/>
     <param name="rtp-timeout-sec" value="300"/>
     <param name="rtp-hold-timeout-sec" value="1800"/>
     <param name="force-register-domain" value="$${domain}"/>
diff --git a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/external.xml b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/vpn.xml
similarity index 84%
rename from modules/rhizo_base/files/etc/freeswitch/sip_profiles/external.xml
rename to modules/rhizo_base/files/etc/freeswitch/sip_profiles/vpn.xml
index cac4952..0f99794 100644
--- a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/external.xml
+++ b/modules/rhizo_base/files/etc/freeswitch/sip_profiles/vpn.xml
@@ -1,8 +1,8 @@
-<profile name="external">
+<profile name="vpn">
   <!-- 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"/>
+    <X-PRE-PROCESS cmd="include" data="gateways/*.xml"/>
   </gateways>
 
   <aliases>
@@ -13,11 +13,14 @@
   </aliases>
 
   <domains>
-    <domain name="all" alias="false" parse="true"/>
+    <domain name="all" alias="false" parse="false"/>
   </domains>
 
   <settings>
-    <param name="NDLB-allow-bad-iananame" value="true"/>
+
+    <param name="auto-jitterbuffer-msec" value="100"/>
+    <param name="minimum-session-expires" value="90"/>
+    <param name="enable-time" value="false"/>
     <param name="debug" value="0"/>
     <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
     <!-- <param name="shutdown-on-fail" value="true"/> -->
@@ -26,12 +29,14 @@
     <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="5059"/>
+    <param name="sip-port" value="5060"/>
     <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="inbound-codec-prefs" value="AMR,GSM,OPUS@8000h@20i,G729,PCMA,PCMU"/>
+    <param name="outbound-codec-prefs" value="AMR,GSM,OPUS@8000h@20i,G729,PCMA,PCMU"/>
+
     <!--<param name="hold-music" value="$${hold_music}"/>-->
     <param name="rtp-timer-name" value="soft"/>
     <!--<param name="enable-100rel" value="true"/>-->
@@ -53,8 +58,8 @@
     <!--<param name="force-register-db-domain" value="$${domain}"/>-->
     <!-- ************************************************* -->
 
-    <param name="aggressive-nat-detection" value="true"/>
-    <param name="nat-options-ping" value="true"/>
+    <param name="aggressive-nat-detection" value="false"/>
+    <param name="nat-options-ping" value="false"/>
     <param name="inbound-codec-negotiation" value="greedy"/>
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="false"/>
@@ -63,10 +68,10 @@
     <!--
         DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
     -->
-    <param name="rtp-ip" value="$${external_ip}"/>
-    <param name="sip-ip" value="$${external_ip}"/>
-    <param name="ext-rtp-ip" value="$${external_ip}"/>
-    <param name="ext-sip-ip" value="$${external_ip}"/>
+    <param name="rtp-ip" value="$${vpn_ip}"/>
+    <param name="sip-ip" value="$${vpn_ip}"/>
+    <param name="ext-rtp-ip" value="$${vpn_ip}"/>
+    <param name="ext-sip-ip" value="$${vpn_ip}"/>
     <param name="rtp-timeout-sec" value="90"/>
     <param name="rtp-hold-timeout-sec" value="1800"/>
     <!--<param name="enable-3pcc" value="true"/>-->
diff --git a/modules/rhizo_base/manifests/freeswitch.pp b/modules/rhizo_base/manifests/freeswitch.pp
index f8ac0c7..959313e 100644
--- a/modules/rhizo_base/manifests/freeswitch.pp
+++ b/modules/rhizo_base/manifests/freeswitch.pp
@@ -173,56 +173,40 @@ class rhizo_base::freeswitch::common {
       require => Package['freeswitch'],
     }
 
-  file {'/etc/freeswitch/sip_profiles/extrtp':
+  file {'/etc/freeswitch/sip_profiles/gateways':
       ensure  => directory,
     }
 
-  file { '/etc/freeswitch/sip_profiles/outgoing.xml':
-      content => template('rhizo_base/outgoing.xml.erb'),
+  file { '/etc/freeswitch/sip_profiles/extrtp.xml':
+      content => template('rhizo_base/extrtp.xml.erb'),
       require =>
                 [ Package[ 'freeswitch' ],
                   File[ '/etc/freeswitch/' ] ],
     }
 
-
-  file { '/etc/freeswitch/sip_profiles/extrtp/rhizoextrtp.xml':
-      content => template('rhizo_base/rhizoextrtp.xml.erb'),
+  file { '/etc/freeswitch/sip_profiles/rem_xcode.xml':
+     content => template('rhizo_base/rem_xcode.xml.erb'),
       require =>
                 [ Package['freeswitch'],
-                File['/etc/freeswitch/sip_profiles/extrtp'] ],
+                File['/etc/freeswitch/'] ],
     }
 
-  file {'/etc/freeswitch/sip_profiles/external':
-      ensure  => directory,
+  file { '/etc/freeswitch/sip_profiles/GERAN.xml':
+      content => template('rhizo_base/GERAN.xml.erb'),
     }
 
-  file { '/etc/freeswitch/sip_profiles/external/provider.xml':
-      content => template('rhizo_base/provider.xml.erb'),
-      require =>
-                [ Package['freeswitch'],
-                File['/etc/freeswitch/sip_profiles/external'] ],
-    }
-
-  file { '/etc/freeswitch/sip_profiles/internal.xml':
-      content => template('rhizo_base/internal.xml.erb'),
-    }
-
-  file {'/etc/freeswitch/sip_profiles/outgoing':
-      ensure  => directory,
-    }
-
-  file { '/etc/freeswitch/sip_profiles/outgoing/rhizomatica.xml':
+  file { '/etc/freeswitch/sip_profiles/gateways/rhizomatica.xml':
       content => template('rhizo_base/rhizomatica.xml.erb'),
       require =>
                 [ Package['freeswitch'],
-                File['/etc/freeswitch/sip_profiles/outgoing'] ],
+                File['/etc/freeswitch/sip_profiles/gateways'] ],
     }
 
-  file { '/etc/freeswitch/sip_profiles/outgoing/rhizoalt.xml':
+  file { '/etc/freeswitch/sip_profiles/gateways/rhizoalt.xml':
       content => template('rhizo_base/rhizoalt.xml.erb'),
       require =>
                 [ Package['freeswitch'],
-                File['/etc/freeswitch/sip_profiles/outgoing'] ],
+                File['/etc/freeswitch/sip_profiles/gateways'] ],
     }
 
   file { '/etc/freeswitch/autoload_configs/acl.conf.xml':
diff --git a/modules/rhizo_base/templates/GERAN.xml.erb b/modules/rhizo_base/templates/GERAN.xml.erb
new file mode 100644
index 0000000..addfa06
--- /dev/null
+++ b/modules/rhizo_base/templates/GERAN.xml.erb
@@ -0,0 +1,191 @@
+<profile name="GERAN">
+
+  <domains>
+    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
+    <!--<domain name="$${domain}" parse="true"/>-->
+    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
+    <!--<domain name="all" alias="true" parse="true"/>-->
+    <domain name="all" alias="false" parse="false"/>
+  </domains>
+
+  <settings>
+    <!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
+    <!-- <param name="rtp-digit-delay" value="40"/>-->
+
+    <!--
+        When calls are in no media this will bring them back to media
+        when you press the hold button.
+    -->
+    <!--<param name="media-option" value="resume-media-on-hold"/> -->
+    <!--
+        This will allow a call after an attended transfer go back to
+        bypass media after an attended transfer.
+    -->
+    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
+    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+    <param name="debug" value="0"/>
+    <!-- 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"/>
+
+    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
+    <!-- <param name="presence-proto-lookup" value="true"/> -->
+
+    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
+    <param name="liberal-dtmf" value="true"/>
+    <param name="watchdog-enabled" value="no"/>
+
+    <param name="log-auth-failures" value="false"/>
+    <param name="forward-unsolicited-mwi-notify" value="false"/>
+
+    <param name="context" value="public"/>
+    <param name="rfc2833-pt" value="101"/>
+    <param name="sip-port" value="5060"/>
+    <param name="dialplan" value="XML"/>
+    <param name="dtmf-duration" value="2000"/>
+    <param name="inbound-codec-prefs" value="GSM,AMR"/>
+    <param name="outbound-codec-prefs" value="GSM,AMR"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <param name="rtp-ip" value="<%= @mncc_ip_address %>"/>
+    <param name="sip-ip" value="<%= @mncc_ip_address %>"/>
+    <!--<param name="hold-music" value="$${hold_music}"/>-->
+    <!-- param name="apply-nat-acl" value="nat.auto"/ -->
+
+    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
+    <!-- <param name="cid-in-1xx" value="false"/> -->
+
+    <!-- extended info parsing -->
+    <!-- <param name="extended-info-parsing" value="true"/> -->
+
+    <param name="aggressive-nat-detection" value="false"/>
+    <!--
+        There are known issues (asserts and segfaults) when 100rel is enabled.
+        It is not recommended to enable 100rel at this time.
+    -->
+    <!--<param name="enable-100rel" value="true"/>-->
+
+    <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
+    <!-- RFC3263 Section 4.3 -->
+    <!--<param name="disable-srv503" value="true"/>-->
+
+    <!-- Enable Compact SIP headers. -->
+    <!--<param name="enable-compact-headers" value="true"/>-->
+    <!--
+        enable/disable session timers
+    -->
+    <!--<param name="enable-timer" value="false"/>-->
+    <!--<param name="minimum-session-expires" value="120"/>-->
+    <param name="apply-inbound-acl" value="domains"/>
+    <!--
+        This defines your local network, by default we detect your local network
+        and create this localnet.auto ACL for this.
+    -->
+    <param name="local-network-acl" value="localnet.auto"/>
+    <param name="apply-register-acl" value="domains"/>
+    <!--<param name="dtmf-type" value="info"/>-->
+
+    <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
+    <!-- Remote-Party-ID header -->
+    <!--<param name="caller-id-type" value="rpid"/>-->
+
+    <!-- P-*-Identity family of headers -->
+    <!--<param name="caller-id-type" value="pid"/>-->
+
+    <!-- neither one -->
+    <!--<param name="caller-id-type" value="none"/>-->
+
+    <param name="record-path" value="$${recordings_dir}"/>
+    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+    <!--enable to use presence -->
+    <param name="manage-presence" value="false"/>
+    <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
+    <!--<param name="presence-probe-on-register" value="true"/>-->
+    <!--<param name="manage-shared-appearance" value="true"/>-->
+    <!-- used to share presence info across sofia profiles -->
+    <!-- Name of the db to use for this profile -->
+    <!--<param name="dbname" value="share_presence"/>-->
+    <param name="presence-hosts" value="$${domain},$${vpn_ip}"/>
+    <param name="presence-privacy" value="$${presence_privacy}"/>
+    <!-- ************************************************* -->
+
+    <!-- This setting is for AAL2 bitpacking on G726 -->
+    <!-- <param name="bitpacking" value="aal2"/> -->
+    <!--max number of open dialogs in proceeding -->
+    <!--<param name="max-proceeding" value="1000"/>-->
+    <!--session timers for all call to expire after the specified seconds -->
+    <!--<param name="session-timeout" value="1800"/>-->
+    <!-- Can be 'true' or 'contact' -->
+    <!--<param name="multiple-registrations" value="contact"/>-->
+    <!--set to 'greedy' if you want your codec list to take precedence -->
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <!-- if you want to send any special bind params of your own -->
+    <!--<param name="bind-params" value="transport=udp"/>-->
+    <!--<param name="unregister-on-options-fail" value="true"/>-->
+    <!-- Send an OPTIONS packet to all registered endpoints -->
+    <!--<param name="all-reg-options-ping" value="true"/>-->
+    <!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
+    <!--<param name="nat-options-ping" value="true"/>-->
+
+    <!-- TLS: disabled by default, set to "true" to enable -->
+    <param name="tls" value="false"/>
+
+    <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
+         (reduces delay on latent connections default true, must be disabled explicitly)-->
+    <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
+
+    <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
+    <param name="rtp-rewrite-timestamps" value="false"/>
+    <!--<param name="pass-rfc2833" value="true"/>-->
+    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+
+    <!--Uncomment to set all inbound calls to no media mode-->
+    <!--<param name="inbound-bypass-media" value="true"/>-->
+
+    <!--Uncomment to set all inbound calls to proxy media mode-->
+    <!--<param name="inbound-proxy-media" value="true"/>-->
+
+    <!-- Let calls hit the dialplan before selecting codec for the a-leg -->
+    <param name="inbound-late-negotiation" value="true"/>
+    <param name="inherit_codec" value="true"/>
+    <!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
+    <!-- param name="inbound-zrtp-passthru" value="true"/ -->
+
+    <!-- this lets anything register -->
+    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+    <!-- <param name="accept-blind-reg" value="true"/> -->
+
+    <!-- accept any authentication without actually checking (not a good feature for most people) -->
+    <!-- <param name="accept-blind-auth" value="true"/> -->
+
+    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
+    <!-- <param name="suppress-cng" value="true"/> -->
+
+    <param name="auth-calls" value="false"/>
+
+    <param name="ext-rtp-ip" value="<%= @mncc_ip_address %>"/>
+    <param name="ext-sip-ip" value="<%= @mncc_ip_address %>"/>
+
+    <!-- rtp inactivity timeout -->
+    <param name="rtp-timeout-sec" value="300"/>
+    <param name="rtp-hold-timeout-sec" value="1800"/>
+    <!-- VAD choose one (out is a good choice); -->
+    <!-- <param name="vad" value="in"/> -->
+    <!-- <param name="vad" value="out"/> -->
+    <!-- <param name="vad" value="both"/> -->
+    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+    <!--
+        These are enabled to make the default config work better out of the box.
+        If you need more than ONE domain you'll need to not use these options.
+
+    -->
+    <!--all inbound reg will look in this domain for the users -->
+    <param name="force-register-domain" value="$${domain}"/>
+    <!--force the domain in subscriptions to this value -->
+    <param name="force-subscription-domain" value="$${domain}"/>
+    <!--all inbound reg will stored in the db using this domain -->
+    <param name="force-register-db-domain" value="$${domain}"/>
+
+  </settings>
+</profile>
diff --git a/modules/rhizo_base/templates/extrtp.xml.erb b/modules/rhizo_base/templates/extrtp.xml.erb
new file mode 100644
index 0000000..accd9db
--- /dev/null
+++ b/modules/rhizo_base/templates/extrtp.xml.erb
@@ -0,0 +1,86 @@
+<profile name="extrtp">
+  <!-- This profile is runs CP over the VPN and media over public -->
+  <!-- It's posssible to have problems with NAT punching -->
+  <gateways>
+    <gateway name="extrtp">
+    <param name="username" value="<%= @voip_username %>"/>
+    <param name="from-user" value="<%= @voip_fromuser %>"/>
+    <param name="password" value="n/a"/>
+    <param name="proxy" value="10.23.0.14"/>
+    <param name="register" value="false"/>
+    <param name="caller-id-in-from" value="true"/>
+    </gateway>
+  </gateways>
+
+  <aliases>
+  </aliases>
+
+  <domains>
+    <domain name="all" alias="false" parse="false"/>
+  </domains>
+
+  <settings>
+    <param name="debug" value="0"/>
+    <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
+    <!-- <param name="shutdown-on-fail" value="true"/> -->
+    <param name="sip-trace" value="no"/>
+    <param name="sip-capture" value="no"/>
+    <param name="rfc2833-pt" value="101"/>
+    <!-- RFC 5626 : Send reg-id and sip.instance -->
+    <!--<param name="enable-rfc-5626" value="true"/> -->
+    <param name="sip-port" value="5092"/>
+    <param name="dialplan" value="XML"/>
+    <param name="context" value="public"/>
+    <param name="dtmf-duration" value="2000"/>
+    <param name="inbound-codec-prefs" value="G729,PCMA,PCMU"/>
+    <param name="outbound-codec-prefs" value="G729,PCMA,PCMU"/>
+    <!--<param name="hold-music" value="$${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"/>
+
+    <param name="aggressive-nat-detection" value="false"/>
+    <param name="nat-options-ping" value="false"/>
+    <param name="inbound-codec-negotiation" value="greedy"/>
+    <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) -->
+
+    <param name="rtp-ip" value="$${interface_ip}"/>
+    <param name="sip-ip" value="$${vpn_ip}"/>
+    <param name="ext-rtp-ip" value="stun:meet.rhizomatica.org:4445"/>
+    <param name="ext-sip-ip" value="$${vpn_ip}"/>
+
+    <param name="rtp-timeout-sec" value="90"/>
+    <param name="rtp-hold-timeout-sec" value="1800"/>
+    <!--<param name="enable-3pcc" value="true"/>-->
+
+    <!-- TLS: disabled by default, set to "true" to enable -->
+    <param name="tls" value="$${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="$${external_tls_port}"/>
+    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+    <!--<param name="tls-cert-dir" value=""/>-->
+    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
+    <param name="tls-passphrase" value=""/>
+    <!-- Verify the date on TLS certificates -->
+    <param name="tls-verify-date" value="true"/>
+    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
+    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
+    <param name="tls-verify-policy" value="none"/>
+    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
+    <param name="tls-verify-depth" value="2"/>
+    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
+    <param name="tls-verify-in-subjects" value=""/>
+    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
+    <param name="tls-version" value="$${sip_tls_version}"/>
+  </settings>
+</profile>
diff --git a/modules/rhizo_base/templates/monitor_freeswitch.sh.erb b/modules/rhizo_base/templates/monitor_freeswitch.sh.erb
index 8aeeb2b..1fb6a1f 100755
--- a/modules/rhizo_base/templates/monitor_freeswitch.sh.erb
+++ b/modules/rhizo_base/templates/monitor_freeswitch.sh.erb
@@ -9,31 +9,32 @@ LOGFILE="/var/log/monitor_fs.log"
 
 IFS=
 FS_STATUS="$(fs_cli -x 'sofia status')"
+FS_STATUS_GW="$(fs_cli -x 'sofia status gateway')"
 
-if !(echo $FS_STATUS | egrep -q "internal.*<%= @mncc_ip_address %>"); then
-	logc "Missing internal profile! Restarting Profile";
-	fs_cli -x "sofia profile internal stop"
+if !(echo $FS_STATUS | egrep -q "GERAN.*<%= @mncc_ip_address %>"); then
+	logc "Missing GERAN profile! Restarting Profile";
+	fs_cli -x "sofia profile GERAN stop"
 	sleep 10
-	fs_cli -x "sofia profile internal start"
+	fs_cli -x "sofia profile GERAN start"
 fi
 
-if !(echo $FS_STATUS | grep -q "external::provider") && (ping -qc 5 8.8.8.8 > /dev/null); then
-	logc "Missing external provider! Restarting Profile";
-	fs_cli -x "sofia profile external stop"
+if !(echo $FS_STATUS_GW | grep -q "extrtp::extrtp") && (ping -qc 5 8.8.8.8 > /dev/null); then
+	logc "Missing external RTP Gateway. Restarting Profile";
+	fs_cli -x "sofia profile extrtp stop"
 	sleep 10
-	fs_cli -x "sofia profile external start"
+	fs_cli -x "sofia profile extrtp start"
 fi
 
-if !(echo $FS_STATUS | grep -q "outgoing::rhizomatica") && (ping -qc 5 10.23.0.2 > /dev/null); then
-	logc "Missing outgoing profile! Restarting Profile";
-	fs_cli -x "sofia profile outgoing stop"
+if !(echo $FS_STATUS_GW | grep -q "vpn::rhizomatica") && (ping -qc 5 10.23.0.2 > /dev/null); then
+	logc "Missing VPN Gateway. Restarting Profile";
+	fs_cli -x "sofia profile vpn stop"
 	sleep 10
-	fs_cli -x "sofia profile outgoing start"
+	fs_cli -x "sofia profile vpn start"
 fi
 
-if !(echo $FS_STATUS | grep -q "internalvpn") && (ping -qc 5 10.23.0.2 > /dev/null); then
-	logc "Missing internal VPN. Restarting FreeSWITCH profile";
-	fs_cli -x "sofia profile internalvpn stop"
+if !(echo $FS_STATUS | grep -q "rem_xcode") && (ping -qc 5 10.23.0.2 > /dev/null); then
+	logc "Missing Remote Transcoding Profile. Restarting FreeSWITCH profile";
+	fs_cli -x "sofia profile rem_xcode stop"
 	sleep 10
-	fs_cli -x "sofia profile internalvpn start"
+	fs_cli -x "sofia profile rem_xcode start"
 fi
diff --git a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/rem_xcode.xml b/modules/rhizo_base/templates/rem_xcode.xml.erb
similarity index 88%
rename from modules/rhizo_base/files/etc/freeswitch/sip_profiles/rem_xcode.xml
rename to modules/rhizo_base/templates/rem_xcode.xml.erb
index 5018358..20fd8ca 100644
--- a/modules/rhizo_base/files/etc/freeswitch/sip_profiles/rem_xcode.xml
+++ b/modules/rhizo_base/templates/rem_xcode.xml.erb
@@ -2,9 +2,9 @@
 
   <gateways>
     <gateway name="xcode">
-    <param name="username" value="rhizomatica"/>
-    <param name="from-user" value="buster"/>
-    <param name="password" value="xcode"/>
+    <param name="username" value="<%= @voip_username %>"/>
+    <param name="from-user" value="<%= @voip_fromuser %>"/>
+    <param name="password" value="n/a"/>
     <param name="proxy" value="10.23.0.14"/>
     <param name="register" value="true"/>
     <param name="caller-id-in-from" value="true"/>
@@ -38,7 +38,7 @@
     <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"/> 
+    <param name="disable-transcoding" value="true"/>
 
     <param name="rtp-ip" value="$${interface_ip}"/>
     <param name="sip-ip" value="$${vpn_ip}"/>
diff --git a/modules/rhizo_base/templates/vars.xml.erb b/modules/rhizo_base/templates/vars.xml.erb
index 80fae6f..55f84d6 100644
--- a/modules/rhizo_base/templates/vars.xml.erb
+++ b/modules/rhizo_base/templates/vars.xml.erb
@@ -38,6 +38,7 @@
 
   <X-PRE-PROCESS cmd="set" data="local_ip_v4=<%= @vpn_ip_address %>"/>
   <X-PRE-PROCESS cmd="set" data="vpn_ip=<%= @vpn_ip_address %>"/>
+  <X-PRE-PROCESS cmd="set" data="lan_ip=<%= @wan_ip_address %>"/>
   <X-PRE-PROCESS cmd="set" data="interface_ip=interface:<%= @fs_bound_if %>"/>
   <X-PRE-PROCESS cmd="set" data="external_ip=<%= @wan_ip_address %>"/>
   <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/rccn"/>
@@ -148,7 +149,7 @@
        127 - BV32
 
   -->
-  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=AMR,OPUS,G729,GSM,PCMA"/>
+  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=AMR,GSM,OPUS,PCMA,PCMU,G729"/>
   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G729,AMR,GSM"/>
 
   <!--