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fun stuff 

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9955 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Brian West 2008-10-11 03:03:46 +00:00
parent 543774ed50
commit 1029aa874f
2 changed files with 3 additions and 1 deletions
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1
conf/dialplan/default.xml
View File

@ -223,6 +223,7 @@
<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>
<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<anti-action application="answer"/>

3
conf/directory/default/brian.xml
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@ -60,7 +60,8 @@
<variable name="effective_caller_id_number" value="1000"/>
<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
<variable name="process_cdr" value="true"/>
<!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="sip_secure_media" value="true"/>
<!-- limit the max number of outgoing calls for this user -->
<!--<variable name="max_calls" value="2"/>-->