Hi Rob, I have adding the missing ">" The wiki is freely editable. You could have made a new login account and signed in and corrected it yourself. Regarding your points 2, 3 -- I think no more edits are required apart from just point 1. Thanks, Sidharth On Thu, Jun 16, 2011 at 7:40 AM, Rob Hutton <justlikeef@gmail.com> wrote:
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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-- Sidharth Kshatriya www.sidk.info