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update default configs slightly

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git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6958 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Brian West 2007-12-22 18:59:36 +00:00
parent d603e22551
commit e7bc12df8f
3 changed files with 8 additions and 66 deletions
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59
conf/dialplan/US.conf.xml
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@ -1,59 +0,0 @@
<context name="US-Numbering-Plan">
<!--
If the destination starts with 011, then check enum and route the call accordingly
-->
<extension name="US_International">
<condition field="destination_number" expression="^011(\d+)$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="enum" data="1$1"/>
<action application="bridge" data="${enum_auto_route}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/011$1"/>
</condition>
</extension>
<!--
If the destination is a 10 or 11 digit US-Style number, then check enum and route accordingly
-->
<extension name="US_LD">
<condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="enum" data="1$1"/>
<action application="bridge" data="${enum_auto_route}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1$1"/>
</condition>
</extension>
<!--
Check for the existence of a default area code.... If found and the destination was dialed
as 7 digits prefix a 1 and the default area code to the dialstring, then check enum,
then (you guessed it) route accordingly
-->
<extension name="US_Local">
<condition field="${default_area_code}" expression="\d{3}" continue="on-true">
<anti-action application="speak" data="you must dial the area code to call this destination"/>
</condition>
<condition field="destination_number" expression="^([2-9]\d{6})$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="enum" data="1$1"/>
<action application="bridge" data="${enum_auto_route}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1${default_area_code}$1"/>
</condition>
</extension>
<!--
If the number dialed was
411 (Directory Services),
611 (Pretty Standard for Provider Customer Service),
811 (call before you dig i think?),
or 911 (Emergency Services) then send directly out the default profile
-->
<extension name="FCC_Services">
<condition field="^([4689]11)$">
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
</condition>
</extension>
</context>

3
conf/dialplan/default.xml
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@ -113,8 +113,7 @@
<anti-action application="set" data="transfer_ringback=${us-ring}"/>
<anti-action application="set" data="call_timeout=30"/>
<anti-action application="set" data="hangup_after_bridge=true"/>
<anti-action application="set" data="left_hanging_extension=5900"/>
<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT"/>
<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
<anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
<anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/>

12
conf/sip_profiles/default.xml
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@ -29,6 +29,8 @@
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${moh_uri}"/>
<!--<param name="dtmf-type" value="info"/>-->
<param name="record-template" value="$${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 -->
@ -44,13 +46,13 @@
<!--<param name="bind-params" value="transport=udp"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<!--<param name="tls" value="false"/>-->
<param name="tls" value="true"/>
<!-- additional bind parameters for TLS -->
<!--<param name="tls-bind-params" value="transport=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"/>-->
<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="$${base_dir}/conf/ssl"/>-->
<param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
<!--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"/>-->
@ -74,7 +76,7 @@
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"/>
<!--<param name="NDLB-broken-auth-hash" 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"/>