[Freeswitch-users] Freeswitch routing inbound calls over SIP instead of TLS/SRTP

[Tim Smith]

Hi Emrah,

Its a test box.  So one internal, one external, one endpoint .... no
scope for confusion. ;-)

On 20 November 2016 at 11:23, Emrah <lists at kavun.ch> wrote:
> Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there.
> How many profiles are you running?
> Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP?
> Also, I went through your message fast, but I don't think you're securing your rtp...
>
>> On Nov 18, 2016, at 11:08 AM, Tim Smith <randomdev4 at gmail.com> wrote:
>>
>> Debian GNU/Linux 8 (jessie)
>> Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19)
>> x86_64 GNU/Linux
>> FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
>>
>> I have a Vtech handset with TLS/SRTP enabled registered with
>> Freeswitch as below:
>>
>>
>> Call-ID:        a0000a0a000aa000
>> User:           2001 at my.example.com
>> Contact:        "my" <sips:2001 at 198.51.100.81:58348>
>> Agent:          Vtech Vesa VSP736A 2.0.3.2-0
>> Status:         Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646)
>> Ping-Status:    Reachable
>> Ping-Time:      0.00
>> Host:           my
>> IP:             198.51.100.81
>> Port:           58348
>> Auth-User:      2001
>> Auth-Realm:     my.example.com
>> MWI-Account:    2001 at my.example.com
>>
>>
>> sofia_contact is happy :
>>
>> freeswitch at my>sofia_contact internal/2001
>> sofia/internal/sip:2001 at 198.51.100.81:58348
>>
>> I have an inbound dial plan configured as follows:
>>
>> <include>
>>  <extension name=“test_inbound">
>>    <condition field="destination_number" expression=“^(15550100)$">
>>     <action application="set" data="domain_name=$${domain}"/>
>>      <action application="bridge" data="${sofia_contact(internal/2001)}"/>
>>    </condition>
>>  </extension>
>> </include>
>>
>> The problem is Freeswitch is sending invites over SIP/RTP and not
>> TLS/SRTP and so the calls never get through :
>>
>> INVITE sip:2001 at 198.51.100.81:58348 SIP/2.0
>>   Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK
>>   From: "Anonymous" <sip:anonymous at 203.0.113.4>;tag=rKmXQjZN8SFXp
>>   To: <sip:2001 at 198.51.100.81:58348>
>>   m=audio 32190 RTP/AVP 8 98 9 101
>>   a=rtpmap:8 PCMA/8000
>>   a=rtpmap:98 G726-32/8000
>>   a=rtpmap:9 G722/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=ptime:20
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org