Hi Rob, Sorry for responding late on this. I'd be happy to help with this - seems like a really good use of time. Want to give a trial run at this over the weekend? - Darren -- On 6/16/11 2:40 PM, "Rob Hutton" <justlikeef@gmail.com> wrote:
I am trying to get encryption working from within Bluebox, in the most "reasonably flexible" way possible. (So no, not the default dialpan, but I missed the example so I will go back and look at it)
So, one scenario I am thinking needs to be supported is where you have two devices that are registered to the same user, one encrypted and one not. For instance, a phone and a remote ringer.
What I am looking for is the best way to stay as flexible as possible. It may be a situation where you end up turning on encryption system wide if the devices support it, but that is overkill in a situation where there is a seperate voice and data VLAN unless there is a need for that level of security..
It may be a situation where I need to offer both options and write two dialplan enries in the situation where the admin wants to enable it device by device.
BTW, I am also using my head to beat through getting TLS working on the front end. I would REALLY appreciate another set of eyes if you have time.
http://jira.freeswitch.org/browse/FS-3346?page=com.atlassian.jira.plugin.s ystem.issuetabpanels:comment- tabpanel&focusedCommentId=24719#action_24719
Thanks, Rob
On Thursday 16 June 2011 14:01:41 Michael Collins wrote:
Are you working off of the default.xml dialplan file? If so, it has an example condition already:
<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"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <!-- <action application="export" data="sip_secure_media=true"/> --> </condition>
What exactly are you checking on in your scenario? Most likely there is an elegant way to do it. Give us the plain language description of the problem you're addressing and the community will no doubt have good suggestions for you.
-MC
On Thu, Jun 16, 2011 at 10:22 AM, Rob Hutton <justlikeef@gmail.com> wrote:
Steven -
Thanks for the help here...
So there would have to be two dialplan entries for this number to work with either RTP or SRTP? (Maybe two devices registering to the same user?)
Would it make more since to do this in a more global manner higher up in the dialplan in its own condition block?
On Thursday 16 June 2011 03:15:33 Steven Ayre wrote:
{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)$" break="never">
<action application="set" data="sip_secure_media=true"/> <action application="bridge" data="openzap/1/1"/>
</condition>
</extension>
The two conditions function as an AND, even though it's not nested. FS stops checking the extension as soon as it sees a condition that's false (at least by default and in the above case), so if the destination is not 202 it'll never get to the 2nd condition.
-Steve
On 16 June 2011 03:10, Rob Hutton <justlikeef@gmail.com> wrote:
I think I have TLS and SRTP working at this point, but in the docs it says to use the following template for the dialplan:
http://wiki.freeswitch.org/wiki/Secure_RTP: <extension name="incoming-fxs">
<condition field="destination_number" expression="^(202)$"/>
<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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