[Freeswitch-svn] [commit] r9314 - freeswitch/trunk/conf/sip_profiles
Freeswitch SVN
brian at freeswitch.org
Sun Aug 17 19:01:57 EDT 2008
Author: brian
Date: Sun Aug 17 19:01:56 2008
New Revision: 9314
Added:
freeswitch/trunk/conf/sip_profiles/internal-ipv6.noload
Log:
example ipv6 enabled profile called internal-ipv6
Added: freeswitch/trunk/conf/sip_profiles/internal-ipv6.noload
==============================================================================
--- (empty file)
+++ freeswitch/trunk/conf/sip_profiles/internal-ipv6.noload Sun Aug 17 19:01:56 2008
@@ -0,0 +1,114 @@
+<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+<profile name="internal-ipv6">
+ <!--aliases are other names that will work as a valid profile name for this profile-->
+ <settings>
+ <!-- <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="5060"/>
+ <param name="dialplan" value="XML"/>
+ <param name="dtmf-duration" value="100"/>
+ <param name="codec-prefs" value="$${global_codec_prefs}"/>
+ <param name="use-rtp-timer" value="true"/>
+ <param name="rtp-timer-name" value="soft"/>
+ <!-- ip address to use for rtp -->
+ <param name="rtp-ip" value="$${local_ip_v6}"/>
+ <!-- ip address to bind to -->
+ <param name="sip-ip" value="$${local_ip_v6}"/>
+ <param name="hold-music" value="$${hold_music}"/>
+ <!--<param name="enable-timer" value="false"/>-->
+ <!--<param name="enable-100rel" value="false"/>-->
+ <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="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+ <!--enable to use presense and mwi -->
+ <param name="manage-presence" value="true"/>
+ <!-- This setting is for AAL2 bitpacking on G726 -->
+ <!-- <param name="bitpacking" value="aal2"/> -->
+ <!--max number of open dialogs in proceeding -->
+ <!--<param name="max-proceeding" value="1000"/>-->
+ <!--session timers for all call to expire after the specified seconds -->
+ <!--<param name="session-timeout" value="120"/>-->
+ <!--<param name="multiple-registrations" value="true"/>-->
+ <!--set to 'greedy' if you want your codec list to take precedence -->
+ <param name="inbound-codec-negotiation" value="generous"/>
+ <!-- if you want to send any special bind params of your own -->
+ <!--<param name="bind-params" value="transport=udp"/>-->
+ <!--<param name="unregister-on-options-fail" value="true"/>-->
+
+ <!-- TLS: disabled by default, set to "true" to enable -->
+ <param name="tls" value="false"/>
+ <!-- additional bind parameters for TLS -->
+ <param name="tls-bind-params" value="transport=tls"/>
+ <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
+ <param name="tls-sip-port" value="5061"/>
+ <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
+ <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
+ <param name="tls-version" value="tlsv1"/>
+
+ <!--If you don't want to pass through timestampes 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"/>-->
+
+ <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
+ <!--<param name="inbound-late-negotiation" 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"/> -->
+
+ <!-- supress CNG on this profile or per call with the 'supress_cng' variable -->
+ <!-- <param name="supress-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="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="cluecon.com"/>-->
+ <!-- disable register and transfer which may be undesirable in a public switch -->
+ <!--<param name="disable-transfer" value="true"/>-->
+ <!--<param name="disable-register" value="true"/>-->
+ <!--<param name="enable-3pcc" value="true"/>-->
+ <!-- use stun when specified (default is true) -->
+ <!--<param name="stun-enabled" value="true"/>-->
+ <!-- use stun when specified (default is true) -->
+ <!-- set to true to have the profile determine stun is not useful and turn it off globally-->
+ <!--<param name="stun-auto-disable" value="true"/>-->
+ </settings>
+</profile>
+
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