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@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
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