forked from Mirrors/freeswitch
188 lines
9.5 KiB
XML
188 lines
9.5 KiB
XML
<!--
|
|
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 -->
|
|
<profile name="internal">
|
|
<!--aliases are other names that will work as a valid profile name for this profile-->
|
|
<aliases>
|
|
<alias name="default"/>
|
|
</aliases>
|
|
<!-- Outbound Registrations -->
|
|
<gateways>
|
|
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
|
|
</gateways>
|
|
|
|
<domains>
|
|
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
|
|
<!--<domain name="$${domain}" parse="true"/>-->
|
|
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
|
|
<!--<domain name="all" alias="true" parse="true"/>-->
|
|
<domain name="all" alias="true" parse="false"/>
|
|
</domains>
|
|
|
|
<settings>
|
|
<!--
|
|
When calls are in no media this will bring them back to media
|
|
when you press the hold button.
|
|
-->
|
|
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
|
<!--
|
|
This will allow a call after an attended transfer go back to
|
|
bypass media after an attended transfer.
|
|
-->
|
|
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
|
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
|
<param name="debug" value="0"/>
|
|
<param name="sip-trace" value="no"/>
|
|
<param name="context" value="public"/>
|
|
<param name="rfc2833-pt" value="101"/>
|
|
<!-- port to bind to for sip traffic -->
|
|
<param name="sip-port" value="$${internal_sip_port}"/>
|
|
<param name="dialplan" value="XML"/>
|
|
<param name="dtmf-duration" value="100"/>
|
|
<param name="codec-prefs" value="$${global_codec_prefs}"/>
|
|
<param name="use-rtp-timer" value="true"/>
|
|
<param name="rtp-timer-name" value="soft"/>
|
|
<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
|
|
<param name="rtp-ip" value="$${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="rfc1918"/>-->
|
|
<!--<param name="aggressive-nat-detection" value="true"/>-->
|
|
<!--<param name="enable-timer" value="false"/>-->
|
|
<!--<param name="enable-100rel" value="true"/>-->
|
|
<!--<param name="minimum-session-expires" value="120"/>-->
|
|
<param name="apply-inbound-acl" value="domains"/>
|
|
<!--<param name="apply-register-acl" value="domains"/>-->
|
|
<!--<param name="dtmf-type" value="info"/>-->
|
|
<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
|
|
<!--enable to use presence -->
|
|
<param name="manage-presence" value="true"/>
|
|
|
|
<!-- used to share presence info across sofia profiles -->
|
|
<!-- Name of the db to use for this profile -->
|
|
<!--<param name="dbname" value="share_presence"/>-->
|
|
<!--<param name="presence-hosts" value="$${domain}"/>-->
|
|
<!-- ************************************************* -->
|
|
|
|
<!-- 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="120"/>-->
|
|
<!-- 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"/>-->
|
|
|
|
<!-- TLS: disabled by default, set to "true" to enable -->
|
|
<param name="tls" value="$${internal_ssl_enable}"/>
|
|
<!-- additional bind parameters for TLS -->
|
|
<param name="tls-bind-params" value="transport=tls"/>
|
|
<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
|
|
<param name="tls-sip-port" value="$${internal_tls_port}"/>
|
|
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
|
|
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
|
<param name="tls-version" value="$${sip_tls_version}"/>
|
|
|
|
<!--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"/>-->
|
|
|
|
<!--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"/>
|
|
|
|
<!-- 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"/>-->
|
|
<!-- 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="$${internal_auth_calls}"/>
|
|
<!-- Force the user and auth-user to match. -->
|
|
<param name="inbound-reg-force-matching-username" value="true"/>
|
|
<!-- on authed calls, authenticate *all* the packets not just invite -->
|
|
<param name="auth-all-packets" value="false"/>
|
|
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
|
|
<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
|
|
<!-- rtp inactivity timeout -->
|
|
<param name="rtp-timeout-sec" value="300"/>
|
|
<param name="rtp-hold-timeout-sec" value="1800"/>
|
|
<!-- VAD choose one (out is a good choice); -->
|
|
<!-- <param name="vad" value="in"/> -->
|
|
<!-- <param name="vad" value="out"/> -->
|
|
<!-- <param name="vad" value="both"/> -->
|
|
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
|
<!--all inbound reg will look in this domain for the users -->
|
|
<!--<param name="force-register-domain" value="$${domain}"/>-->
|
|
<!--all inbound reg will stored in the db using this domain -->
|
|
<!--<param name="force-register-db-domain" value="$${domain}"/>-->
|
|
<!--force suscription expires to a lower value than requested-->
|
|
<!--<param name="force-subscription-expires" value="60"/>-->
|
|
<!-- 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"/>-->
|
|
</settings>
|
|
</profile>
|