I think I have TLS and SRTP working at this point, but in the docs it says to use the following template for the dialplan:
http://wiki.freeswitch.org/wiki/Secure_RTP:
<extension name="incoming-fxs">
<condition field="destination_number" expression="^(202)$"/>
<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
<action application="set" data="sip_secure_media=true"/>
<action application="bridge" data="openzap/1/1"/>
</condition>
</extension
1) There is a missing > at the end of the close extension tag.
2) There is either a missing / at the end of the internal condition line, or a missing condition close tag somewhere
3) When I fix the interal condition, I get an error:
{ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed
All this, but a packet capture shows that SRTP is working based on what I did on:
http://wiki.freeswitch.org/wiki/SIP_TLS
Can someone give me some guidance on the Secure_RTP page and I will update whatever?
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