[Freeswitch-users] FS with SSL/TLS issues!

[Vittorio Guglielmo]

I did it,
now I have:

/mod_dptools.c:3052 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION/

The only way I succeded was to add /bypass_media=true/ in the dialplan
for each extension,
but I have a huge latency in the voice (like 2/3 seconds).


Vic



On 27/02/13 22:54, Anthony Minessale wrote:
> This will work better in 1.4 but the code in 1.2 is frozen so we can't
> improve it there.
> In the meantime you can just do this: 
>
> make an acl that is default allow called tls
> add it to apply-nat-acl param in sofia profile where you have tls
>
> Now it will record the rport for every client in the contact path and
> it should resolve the existing connection.
>
>
>
>
> On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo
> <vittorio.guglielmo at gmail.com <mailto:vittorio.guglielmo at gmail.com>>
> wrote:
>
>     Hi,
>
>     I've done all you said but I'm in trouble indeed:
>
>     TLS/SRTP enabled on two extension;
>
>     If I try to dial between the two extensions I have:
>
>     /mod_dptools.c:3052 Originate Failed.  Cause:
>     INCOMPATIBLE_DESTINATION/
>
>     If I call outbound, it's ok, and the call is encrypted.
>
>     Can you help me?
>
>     Vic
>
>
>
>
>     On 27/02/13 15:47, mehroz wrote:
>>     TLS-NAT issue arises due to the fact that it tries to contact on other port
>>     (TCP session) as far as my understanding.
>>
>>     Changing contact header can sort this issue and i have done it setting up
>>
>>     <variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/>
>>     for each directory (user).
>>
>>     This helped and worked :)
>>
>>     I suppose you should also have "ext-rtp-ip and "ext-sip-ip" =  "nat-auto" 
>>     in SIP profile.
>>     Cheeers!
>>
>>
>>
>>     --
>>     View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587736p7588069.html
>>     Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
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