Hi Rob,<br><br>I have adding the missing ">"<br><br>The wiki is freely editable. You could have made a new login account and signed in and corrected it yourself.<br><br>Regarding your points 2, 3 -- I think no more edits are required apart from just point 1.<br>
<br>Thanks,<br><br>Sidharth<br><br><div class="gmail_quote">On Thu, Jun 16, 2011 at 7:40 AM, Rob Hutton <span dir="ltr"><<a href="mailto:justlikeef@gmail.com">justlikeef@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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:<br>
<br>
<a href="http://wiki.freeswitch.org/wiki/Secure_RTP" target="_blank">http://wiki.freeswitch.org/wiki/Secure_RTP</a>:<br>
<br>
<extension name="incoming-fxs"><br>
<condition field="destination_number" expression="^(202)$"/><br>
<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"><br>
<action application="set" data="sip_secure_media=true"/><br>
<action application="bridge" data="openzap/1/1"/><br>
</condition><br>
</extension<br>
<br>
1) There is a missing > at the end of the close extension tag.<br>
2) There is either a missing / at the end of the internal condition line, or a missing condition close tag somewhere<br>
3) When I fix the interal condition, I get an error:<br>
<br>
{ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed<br>
<br>
All this, but a packet capture shows that SRTP is working based on what I did on:<br>
<br>
<a href="http://wiki.freeswitch.org/wiki/SIP_TLS" target="_blank">http://wiki.freeswitch.org/wiki/SIP_TLS</a><br>
<br>
Can someone give me some guidance on the Secure_RTP page and I will update whatever?<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Sidharth Kshatriya<br><a href="http://www.sidk.info">www.sidk.info</a><br><br>