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FS-5257 --resolve you seem to be setting sip_secure_media on an inbound leg, this is not correct. This patch should auto-correct that but also beware that the variable has changed to rtp_secure_media after this commit
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@@ -95,9 +95,9 @@
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<action application="info"/>
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</condition>
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<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
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<action application="set" data="sip_secure_media=true"/>
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<action application="set" data="rtp_secure_media=true"/>
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<!-- Offer SRTP on outbound legs if we have it on inbound. -->
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<!-- <action application="export" data="sip_secure_media=true"/> -->
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<!-- <action application="export" data="rtp_secure_media=true"/> -->
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</condition>
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<condition>
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<action application="db" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
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@@ -231,7 +231,7 @@
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<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
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<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
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<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
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<!--<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
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<!--<anti-action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
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<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
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<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
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<anti-action application="answer"/>
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@@ -40,7 +40,7 @@
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<extension name="is_secure">
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<!-- Only Truly consider it secure if its TLS and SRTP -->
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<condition field="${sip_via_protocol}" expression="tls"/>
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<condition field="${sip_secure_media_confirmed}" expression="^true$">
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<condition field="${rtp_secure_media_confirmed}" expression="^true$">
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<action application="sleep" data="1000"/>
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<action application="playback" data="misc/call_secured.wav"/>
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<anti-action application="eval" data="not_secure"/>
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@@ -60,8 +60,8 @@
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<variable name="effective_caller_id_number" value="1000"/>
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<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
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<variable name="process_cdr" value="true"/>
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<!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
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<variable name="sip_secure_media" value="true"/>
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<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
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<variable name="rtp_secure_media" value="true"/>
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<!-- limit the max number of outgoing calls for this user -->
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<!--<variable name="max_calls" value="2"/>-->
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@@ -95,9 +95,9 @@
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<action application="info"/>
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</condition>
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<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
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<action application="set" data="sip_secure_media=true"/>
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<action application="set" data="rtp_secure_media=true"/>
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<!-- Offer SRTP on outbound legs if we have it on inbound. -->
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<!-- <action application="export" data="sip_secure_media=true"/> -->
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<!-- <action application="export" data="rtp_secure_media=true"/> -->
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</condition>
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<condition>
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<action application="db" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
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@@ -231,7 +231,7 @@
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<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
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<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
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<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
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<!--<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
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<!--<anti-action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
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<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
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<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
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<anti-action application="answer"/>
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@@ -40,7 +40,7 @@
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<extension name="is_secure">
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<!-- Only Truly consider it secure if its TLS and SRTP -->
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<condition field="${sip_via_protocol}" expression="tls"/>
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<condition field="${sip_secure_media_confirmed}" expression="^true$">
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<condition field="${rtp_secure_media_confirmed}" expression="^true$">
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<action application="sleep" data="1000"/>
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<action application="playback" data="misc/call_secured.wav"/>
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<anti-action application="eval" data="not_secure"/>
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@@ -136,9 +136,9 @@
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</condition>
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-->
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<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
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<action application="set" data="sip_secure_media=true"/>
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<action application="set" data="rtp_secure_media=true"/>
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<!-- Offer SRTP on outbound legs if we have it on inbound. -->
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<!-- <action application="export" data="sip_secure_media=true"/> -->
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<!-- <action application="export" data="rtp_secure_media=true"/> -->
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</condition>
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<!--
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@@ -147,9 +147,9 @@
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-->
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<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/>
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<condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never">
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<action application="set" data="sip_secure_media=true"/>
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<action application="set" data="rtp_secure_media=true"/>
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<!-- Offer SRTP on outbound legs if we have it on inbound. -->
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<!-- <action application="export" data="sip_secure_media=true"/> -->
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<!-- <action application="export" data="rtp_secure_media=true"/> -->
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</condition>
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@@ -274,7 +274,7 @@
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<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
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<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
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<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
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<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
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<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
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<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
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<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
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<action application="answer"/>
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@@ -56,7 +56,7 @@
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<extension name="is_secure" continue="true">
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<!-- Only Truly consider it secure if its TLS and SRTP -->
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<condition field="${sip_via_protocol}" expression="tls"/>
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<condition field="${sip_secure_media_confirmed}" expression="^true$">
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<condition field="${rtp_secure_media_confirmed}" expression="^true$">
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<action application="sleep" data="1000"/>
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<action application="playback" data="misc/call_secured.wav"/>
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<anti-action application="eval" data="not_secure"/>
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@@ -62,8 +62,8 @@
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<variable name="effective_caller_id_number" value="1000"/>
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<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
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<variable name="process_cdr" value="true"/>
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<!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
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<variable name="sip_secure_media" value="true"/>
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<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
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<variable name="rtp_secure_media" value="true"/>
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<!-- limit the max number of outgoing calls for this user -->
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<!--<variable name="max_calls" value="2"/>-->
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