FS: Rename SIP Profiles
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0e24b1e31b
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1fbe4aef12
10 changed files with 333 additions and 514 deletions
191
modules/rhizo_base/templates/GERAN.xml.erb
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191
modules/rhizo_base/templates/GERAN.xml.erb
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@ -0,0 +1,191 @@
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<profile name="GERAN">
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<domains>
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<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
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<!--<domain name="$${domain}" parse="true"/>-->
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<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
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<!--<domain name="all" alias="true" parse="true"/>-->
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<domain name="all" alias="false" parse="false"/>
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</domains>
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<settings>
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<!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
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<!-- <param name="rtp-digit-delay" value="40"/>-->
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<!--
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When calls are in no media this will bring them back to media
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when you press the hold button.
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-->
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<!--<param name="media-option" value="resume-media-on-hold"/> -->
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<!--
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This will allow a call after an attended transfer go back to
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bypass media after an attended transfer.
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-->
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<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
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<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
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<param name="debug" value="0"/>
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<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
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<param name="shutdown-on-fail" value="true"/>
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<param name="sip-trace" value="no"/>
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<param name="sip-capture" value="no"/>
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<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
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<!-- <param name="presence-proto-lookup" value="true"/> -->
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<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
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<param name="liberal-dtmf" value="true"/>
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<param name="watchdog-enabled" value="no"/>
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<param name="log-auth-failures" value="false"/>
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<param name="forward-unsolicited-mwi-notify" value="false"/>
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<param name="context" value="public"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="GSM,AMR"/>
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<param name="outbound-codec-prefs" value="GSM,AMR"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="rtp-ip" value="<%= @mncc_ip_address %>"/>
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<param name="sip-ip" value="<%= @mncc_ip_address %>"/>
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<!--<param name="hold-music" value="$${hold_music}"/>-->
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<!-- param name="apply-nat-acl" value="nat.auto"/ -->
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<!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
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<!-- <param name="cid-in-1xx" value="false"/> -->
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<!-- extended info parsing -->
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<!-- <param name="extended-info-parsing" value="true"/> -->
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<param name="aggressive-nat-detection" value="false"/>
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<!--
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There are known issues (asserts and segfaults) when 100rel is enabled.
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It is not recommended to enable 100rel at this time.
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-->
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<!--<param name="enable-100rel" value="true"/>-->
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<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
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<!-- RFC3263 Section 4.3 -->
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<!--<param name="disable-srv503" value="true"/>-->
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<!-- Enable Compact SIP headers. -->
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<!--<param name="enable-compact-headers" value="true"/>-->
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<!--
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enable/disable session timers
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-->
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<!--<param name="enable-timer" value="false"/>-->
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<!--<param name="minimum-session-expires" value="120"/>-->
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<param name="apply-inbound-acl" value="domains"/>
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<!--
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This defines your local network, by default we detect your local network
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and create this localnet.auto ACL for this.
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-->
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="apply-register-acl" value="domains"/>
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<!--<param name="dtmf-type" value="info"/>-->
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<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
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<!-- Remote-Party-ID header -->
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<!--<param name="caller-id-type" value="rpid"/>-->
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<!-- P-*-Identity family of headers -->
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<!--<param name="caller-id-type" value="pid"/>-->
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<!-- neither one -->
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<!--<param name="caller-id-type" value="none"/>-->
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<param name="record-path" value="$${recordings_dir}"/>
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<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
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<!--enable to use presence -->
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<param name="manage-presence" value="false"/>
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<!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
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<!--<param name="presence-probe-on-register" value="true"/>-->
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<!--<param name="manage-shared-appearance" value="true"/>-->
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<!-- used to share presence info across sofia profiles -->
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<!-- Name of the db to use for this profile -->
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<!--<param name="dbname" value="share_presence"/>-->
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<param name="presence-hosts" value="$${domain},$${vpn_ip}"/>
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<param name="presence-privacy" value="$${presence_privacy}"/>
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<!-- ************************************************* -->
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<!-- This setting is for AAL2 bitpacking on G726 -->
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<!-- <param name="bitpacking" value="aal2"/> -->
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<!--max number of open dialogs in proceeding -->
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<!--<param name="max-proceeding" value="1000"/>-->
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<!--session timers for all call to expire after the specified seconds -->
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<!--<param name="session-timeout" value="1800"/>-->
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<!-- Can be 'true' or 'contact' -->
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<!--<param name="multiple-registrations" value="contact"/>-->
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<!--set to 'greedy' if you want your codec list to take precedence -->
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<param name="inbound-codec-negotiation" value="generous"/>
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<!-- if you want to send any special bind params of your own -->
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<!--<param name="bind-params" value="transport=udp"/>-->
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<!--<param name="unregister-on-options-fail" value="true"/>-->
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<!-- Send an OPTIONS packet to all registered endpoints -->
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<!--<param name="all-reg-options-ping" value="true"/>-->
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<!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
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<!--<param name="nat-options-ping" value="true"/>-->
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="false"/>
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<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
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(reduces delay on latent connections default true, must be disabled explicitly)-->
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<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
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<!--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)-->
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<param name="rtp-rewrite-timestamps" value="false"/>
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!-- Let calls hit the dialplan before selecting codec for the a-leg -->
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<param name="inbound-late-negotiation" value="true"/>
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<param name="inherit_codec" value="true"/>
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<!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
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<!-- param name="inbound-zrtp-passthru" value="true"/ -->
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<param name="auth-calls" value="false"/>
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<param name="ext-rtp-ip" value="<%= @mncc_ip_address %>"/>
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<param name="ext-sip-ip" value="<%= @mncc_ip_address %>"/>
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<!-- rtp inactivity timeout -->
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<param name="rtp-timeout-sec" value="300"/>
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<param name="rtp-hold-timeout-sec" value="1800"/>
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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<!--
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These are enabled to make the default config work better out of the box.
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If you need more than ONE domain you'll need to not use these options.
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-->
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<!--all inbound reg will look in this domain for the users -->
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<param name="force-register-domain" value="$${domain}"/>
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<!--force the domain in subscriptions to this value -->
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<param name="force-subscription-domain" value="$${domain}"/>
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<!--all inbound reg will stored in the db using this domain -->
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<param name="force-register-db-domain" value="$${domain}"/>
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</settings>
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</profile>
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86
modules/rhizo_base/templates/extrtp.xml.erb
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86
modules/rhizo_base/templates/extrtp.xml.erb
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@ -0,0 +1,86 @@
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<profile name="extrtp">
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<!-- This profile is runs CP over the VPN and media over public -->
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<!-- It's posssible to have problems with NAT punching -->
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<gateways>
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<gateway name="extrtp">
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<param name="username" value="<%= @voip_username %>"/>
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<param name="from-user" value="<%= @voip_fromuser %>"/>
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<param name="password" value="n/a"/>
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<param name="proxy" value="10.23.0.14"/>
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<param name="register" value="false"/>
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<param name="caller-id-in-from" value="true"/>
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</gateway>
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</gateways>
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<aliases>
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</aliases>
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<domains>
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<domain name="all" alias="false" parse="false"/>
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</domains>
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<settings>
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<param name="debug" value="0"/>
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<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
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<!-- <param name="shutdown-on-fail" value="true"/> -->
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<param name="sip-trace" value="no"/>
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<param name="sip-capture" value="no"/>
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<param name="rfc2833-pt" value="101"/>
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<!-- RFC 5626 : Send reg-id and sip.instance -->
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<!--<param name="enable-rfc-5626" value="true"/> -->
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<param name="sip-port" value="5092"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="G729,PCMA,PCMU"/>
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<param name="outbound-codec-prefs" value="G729,PCMA,PCMU"/>
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<!--<param name="hold-music" value="$${hold_music}"/>-->
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<param name="rtp-timer-name" value="soft"/>
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<!--<param name="enable-100rel" value="true"/>-->
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<!--<param name="disable-srv503" value="true"/>-->
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<!-- This could be set to "passive" -->
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="manage-presence" value="false"/>
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<param name="aggressive-nat-detection" value="false"/>
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<param name="nat-options-ping" value="false"/>
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<param name="inbound-codec-negotiation" value="greedy"/>
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<param name="nonce-ttl" value="60"/>
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<param name="auth-calls" value="false"/>
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<param name="inbound-late-negotiation" value="true"/>
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<param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
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<param name="rtp-ip" value="$${interface_ip}"/>
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<param name="sip-ip" value="$${vpn_ip}"/>
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<param name="ext-rtp-ip" value="stun:meet.rhizomatica.org:4445"/>
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<param name="ext-sip-ip" value="$${vpn_ip}"/>
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<param name="rtp-timeout-sec" value="90"/>
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<param name="rtp-hold-timeout-sec" value="1800"/>
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<!--<param name="enable-3pcc" value="true"/>-->
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${external_ssl_enable}"/>
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<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
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<param name="tls-only" value="false"/>
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<!-- additional bind parameters for TLS -->
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<param name="tls-bind-params" value="transport=tls"/>
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<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
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<param name="tls-sip-port" value="$${external_tls_port}"/>
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<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
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<!--<param name="tls-cert-dir" value=""/>-->
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<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
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<param name="tls-passphrase" value=""/>
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<!-- Verify the date on TLS certificates -->
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<param name="tls-verify-date" value="true"/>
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<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
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<!-- 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 -->
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<param name="tls-verify-policy" value="none"/>
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<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
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<param name="tls-verify-depth" value="2"/>
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<!-- 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 -->
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<param name="tls-verify-in-subjects" value=""/>
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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</settings>
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</profile>
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@ -9,31 +9,32 @@ LOGFILE="/var/log/monitor_fs.log"
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IFS=
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FS_STATUS="$(fs_cli -x 'sofia status')"
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FS_STATUS_GW="$(fs_cli -x 'sofia status gateway')"
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if !(echo $FS_STATUS | egrep -q "internal.*<%= @mncc_ip_address %>"); then
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logc "Missing internal profile! Restarting Profile";
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fs_cli -x "sofia profile internal stop"
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if !(echo $FS_STATUS | egrep -q "GERAN.*<%= @mncc_ip_address %>"); then
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logc "Missing GERAN profile! Restarting Profile";
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fs_cli -x "sofia profile GERAN stop"
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sleep 10
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fs_cli -x "sofia profile internal start"
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fs_cli -x "sofia profile GERAN start"
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fi
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if !(echo $FS_STATUS | grep -q "external::provider") && (ping -qc 5 8.8.8.8 > /dev/null); then
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logc "Missing external provider! Restarting Profile";
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fs_cli -x "sofia profile external stop"
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if !(echo $FS_STATUS_GW | grep -q "extrtp::extrtp") && (ping -qc 5 8.8.8.8 > /dev/null); then
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logc "Missing external RTP Gateway. Restarting Profile";
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fs_cli -x "sofia profile extrtp stop"
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sleep 10
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fs_cli -x "sofia profile external start"
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fs_cli -x "sofia profile extrtp start"
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fi
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if !(echo $FS_STATUS | grep -q "outgoing::rhizomatica") && (ping -qc 5 10.23.0.2 > /dev/null); then
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logc "Missing outgoing profile! Restarting Profile";
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fs_cli -x "sofia profile outgoing stop"
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if !(echo $FS_STATUS_GW | grep -q "vpn::rhizomatica") && (ping -qc 5 10.23.0.2 > /dev/null); then
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logc "Missing VPN Gateway. Restarting Profile";
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fs_cli -x "sofia profile vpn stop"
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sleep 10
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fs_cli -x "sofia profile outgoing start"
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fs_cli -x "sofia profile vpn start"
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fi
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if !(echo $FS_STATUS | grep -q "internalvpn") && (ping -qc 5 10.23.0.2 > /dev/null); then
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logc "Missing internal VPN. Restarting FreeSWITCH profile";
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fs_cli -x "sofia profile internalvpn stop"
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if !(echo $FS_STATUS | grep -q "rem_xcode") && (ping -qc 5 10.23.0.2 > /dev/null); then
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logc "Missing Remote Transcoding Profile. Restarting FreeSWITCH profile";
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fs_cli -x "sofia profile rem_xcode stop"
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sleep 10
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fs_cli -x "sofia profile internalvpn start"
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fs_cli -x "sofia profile rem_xcode start"
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fi
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52
modules/rhizo_base/templates/rem_xcode.xml.erb
Normal file
52
modules/rhizo_base/templates/rem_xcode.xml.erb
Normal file
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@ -0,0 +1,52 @@
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<profile name="rem_xcode">
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<gateways>
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<gateway name="xcode">
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<param name="username" value="<%= @voip_username %>"/>
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<param name="from-user" value="<%= @voip_fromuser %>"/>
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<param name="password" value="n/a"/>
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<param name="proxy" value="10.23.0.14"/>
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<param name="register" value="true"/>
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<param name="caller-id-in-from" value="true"/>
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</gateway>
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</gateways>
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<aliases>
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</aliases>
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<domains>
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<domain name="all" alias="false" parse="false"/>
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</domains>
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<settings>
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<param name="debug" value="0"/>
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<param name="sip-trace" value="no"/>
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<param name="sip-capture" value="no"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="5062"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="AMR,GSM"/>
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<param name="outbound-codec-prefs" value="AMR,GSM"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="manage-presence" value="false"/>
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<param name="aggressive-nat-detection" value="false"/>
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<param name="nat-options-ping" value="false"/>
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<param name="inbound-codec-negotiation" value="greedy"/>
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<param name="nonce-ttl" value="60"/>
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<param name="auth-calls" value="false"/>
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<param name="inbound-late-negotiation" value="true"/>
|
||||
<param name="disable-transcoding" value="true"/>
|
||||
|
||||
<param name="rtp-ip" value="$${interface_ip}"/>
|
||||
<param name="sip-ip" value="$${vpn_ip}"/>
|
||||
<param name="ext-rtp-ip" value="stun:sat.rhizomatica.org:3479"/>
|
||||
<param name="ext-sip-ip" value="$${vpn_ip}"/>
|
||||
<param name="rtp-timeout-sec" value="90"/>
|
||||
<param name="rtp-hold-timeout-sec" value="1800"/>
|
||||
|
||||
<param name="tls" value="false"/>
|
||||
</settings>
|
||||
</profile>
|
||||
|
|
@ -38,6 +38,7 @@
|
|||
|
||||
<X-PRE-PROCESS cmd="set" data="local_ip_v4=<%= @vpn_ip_address %>"/>
|
||||
<X-PRE-PROCESS cmd="set" data="vpn_ip=<%= @vpn_ip_address %>"/>
|
||||
<X-PRE-PROCESS cmd="set" data="lan_ip=<%= @wan_ip_address %>"/>
|
||||
<X-PRE-PROCESS cmd="set" data="interface_ip=interface:<%= @fs_bound_if %>"/>
|
||||
<X-PRE-PROCESS cmd="set" data="external_ip=<%= @wan_ip_address %>"/>
|
||||
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${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"/>
|
||||
|
||||
<!--
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue