As i understand you try overwrite transport to user B registration.
In many case users is located behind NAT and FS cannot establish TLS connections to B-user.

Think in your case need to disable all non TLS sockets and then simple try bridge "user/{user}@{domain}"

сб, 29 сент. 2018 г. в 13:20, Chhorm Chhatra <ch.chhatra@gmail.com>:

Dear Brain West,
thank you for your response.

I would like to confirm that either using export or set on a leg of "rtp_secure_media=true" with the following dial-string is not working for me. One leg call is fine but it does not work for 2-leg call (I could not hear the sound and the call terminates after 

  {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" 

On Wed, 1 Aug 2018 at 23:20, Brian West <brian@freeswitch.com> wrote:

don't us export, set it inside {}, or on use set on a-leg.

/b

On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra <ch.chhatra@gmail.com> wrote:

Hello,

Currently, I faced a problem regarding SRTP outbound call to user (Leg B).

The scenario is like this,

• We set up our own root CA to an IP address (e.g 192.168.0.13)
• We create a server certificate for freeswitch at 192.168.0.13
• Linphone is used as SIP client and is configured to trust our root CA by default.
• Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure)
• Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to.

(One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs)

• Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes SRTP unprotect failed with code 7 (auth check failed).

+ Dialplan Configuration

<action application="set" data="rtp_secure_media=true"/>

<action application="export" data="rtp_secure_media=true"/>

The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

+ Directory Configruation:

<param name="dial-string" value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" />

My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)?

Any help would be really appreciated.

Thank you so much. 

Best Regard,

_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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https://freeswitch.com

_________________________________________________________________________
Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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https://freeswitch.com