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

[Steven Ayre]

Looks like a bug to me. Your first snippet shows the contact stored in the
database uses the 'sips:' scheme, but sofia_contact is returning 'sip:'

In the code it looks like sofia_contact fetches the contact
using select_from_profile which invokes contact_callback.
In contact_callback it's hardcoded to use sip: plus the result
of sofia_glue_strip_proto. That looks to me like it can never return a sips
URI even though it's stored in the database.

I'd file a jira.

Steve

On 18 November 2016 at 10:08, 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
>
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