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@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!



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