[Freeswitch-users] SRTP

[Sidharth Kshatriya]

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 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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-- 
Sidharth Kshatriya
www.sidk.info
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