tweak default conf for better first experience

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@4071 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale 2007-01-28 03:05:20 +00:00
parent 781c74fc8e
commit 066807ed52
3 changed files with 6 additions and 6 deletions

View File

@ -29,7 +29,7 @@
<!-- Call the FreeSWITCH conference via SIP -->
<!--<extension name="FreeSWITCH Conference SIP">-->
<!--<condition field="destination_number" expression="^888$">-->
<!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
<!--<action application="bridge" data="sofia/$${domain}/888@conference.freeswitch.org"/>-->
<!--</condition>-->
<!--</extension> -->
@ -45,7 +45,7 @@
<!-- Request a certain tone/file to be played while you wait for the call to be answered-->
<action application="set" data="ringback=${us-ring}"/>
<!--<action application="set" data="ringback=/home/ring.wav"/>-->
<action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
<action application="bridge" data="sofia/$${domain}/1234@conference.freeswitch.org"/>
</condition>
</extension>
@ -59,7 +59,7 @@
<!-- Start a dynamic conference and call someone at the same time -->
<extension name="2000">
<condition field="destination_number" expression="^2000$">
<action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
<action application="conference" data="bridge:mydynaconf:sofia/$${domain}/1234@conference.freeswitch.org"/>
</condition>
</extension>

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@ -4,8 +4,8 @@
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${domain}/$1"/>
<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${domain}/$1"/>
</routes>
</configuration>

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@ -17,7 +17,7 @@
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dialplan" value="enum,XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${default_codecs}"/>
<param name="codec-ms" value="20"/>