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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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Anthony Minessale committed 2013-04-02 20:05:46 -05:00
1 parent 5544db8d5e
commit 69c3c7d8e2
11 files changed
+37 -22
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conf/insideout/directory/default/brian.xml
+2 -2
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@@ -60,8 +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"/>
<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="rtp_secure_media" value="true"/>
<!-- limit the max number of outgoing calls for this user -->
<!--<variable name="max_calls" value="2"/>-->