[Freeswitch-users] Call from Secure RTP to non-secure RTP

[Jim Burke]

Does 1002 use TLS to transport SIP signalling? My experience is that  
TLS is required on some phones otherwise they will not do srtp and  
will reply with the responce you have mentioned.

Sent from my iPhone

On 19/11/2009, at 1:36 PM, Mark Campbell-Smith  
<mcampbellsmith at gmail.com> wrote:

> Hi!
>
> How do I setup FS so that placing a call from an extension that only
> support SRTP (1002) to an extension that only supports RTP (1000)?
>
> I put this dialstring, from the wiki
> http://wiki.freeswitch.org/wiki/Tls, into the users xml file under
> directory/default
>
> <param name="dial-string"
> value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@$ 
> {dialed_domain})}|transport=tls)},
>  presence_id=${dialed_user}@${dialed_domain}}${sofia_contact($ 
> {dialed_user}@${dialed_domain})}"
> />
>
> I have also put a <action application="export"
> data="sip_secure_media=true"/> when 1000 is dialing 1002.
>      <condition field="destination_number" expression="^(1002)$">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="export" data="sip_secure_media=true"/>
>        <action application="bridge" data="user/${dialed_extension}@$ 
> {domain}"/>
>
> However I never see crytpo sent in the RTP to 1002 and it responds
> with Bad Security Level
>
> What have I missed?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org