FS: Rename SIP Profiles
This commit is contained in:
parent
0e24b1e31b
commit
1fbe4aef12
10 changed files with 333 additions and 514 deletions
191
modules/rhizo_base/templates/GERAN.xml.erb
Normal file
191
modules/rhizo_base/templates/GERAN.xml.erb
Normal file
|
|
@ -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>
|
||||
Loading…
Add table
Add a link
Reference in a new issue