вс, 21 окт. 2018 г. в 21:12, Chhorm Chhatra <ch.chhatra@gmail.com>:
Thank you Safarov for your solution.I am not quite sure about reverting the dial-string part.Could you please electorate on how can I revert the dial-string?Do I need to remove the dial-string from the dialplan or from the user directory configuration?Thank you in advance._________________________________________________________________________On Sat, Sep 29, 2018 at 10:52 PM Sergey Safarov <s.safarov@gmail.com> wrote:Need to revert back "dial-string" in directory config
Also important "sips" and "sip" uri different. Please make sure you not use sips uri in client side.Sergeyсб, 29 сент. 2018 г. в 13:35, Sergey Safarov <s.safarov@gmail.com>:_________________________________________________________________________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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