[Freeswitch-users] SRTP

Steven Ayre steveayre at gmail.com
Thu Jun 16 11:15:33 MSD 2011


 {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed

That's because it shouldn't be nested. It's not missing a /, and the 1st
Should have the /. The extra indendation shouldn't be there on the 2nd.

It should look like this:

<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>

The two conditions function as an AND, even though it's not nested. FS stops
checking the extension as soon as it sees a condition that's false (at least
by default and in the above case), so if the destination is not 202 it'll
never get to the 2nd condition.

-Steve



On 16 June 2011 03:10, Rob Hutton <justlikeef at 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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