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

[Tim Smith]

Hi Steve,

Sorry for the delay ack'ing your mail ... yeah, guess I should maybe
look into filing a JIRA.

On 20 November 2016 at 00:33, Steven Ayre <steveayre at gmail.com> wrote:
> 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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>> http://www.freeswitchsolutions.com
>>
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>>
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>
>
>
> _________________________________________________________________________
> 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
>
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