[Freeswitch-users] T.38 gateway and peer/self

Brian : brians at iptel.co
Fri Feb 24 16:21:30 MSK 2017


Hi Brian/Peter

Sorry to jump in on your thread!

Brian I have a use case that I'm having some trouble with - what do I need
to do in this scenario:

Carrier sends call to FS (g711)
FS sends call to endpoint (g711)
endpoint answers then quickly sends a T38 (RE)INVITE back to FS

So in this case we would like to accept this customer (re)invite - do T38
with the customer endpoint and transcode that to 711 back to the A leg with
the carrier.

I've tried every combination of dialplan / parameters from the T38 bits at
https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp and they
mostly result in FS sending a not acceptable here back to the T38 invite.


Thanks,
Brian



On Fri, Feb 24, 2017 at 7:52 AM, Peter Steinbach <lists at telefaks.de> wrote:

> Hello Brian,
>
> this Version of Freeswitch is from 09-Feb-2016. So I think, we should
> update.
>
> We need to cover 2 scenarios
>
>    - send fax to SIP Provider(b leg) and have T.38 Re-invite from SIP
>    Provider and pass it to a-leg
>    - send fax to SIP Provider(b leg), do NOT receive T.38 Re-invite from
>    SIP Provider and then do T.38 Gateway in FS.
>
> And no: We cannot foresee, whether the SIp provider will do T.38 and not.
> Here in Germany, the main incumbant PSTN provider did not implement T.38,
> so it depends on the target number and actual route, if this will be T.38
> or not. A call to a certain number can be T.38 today and tomorrow it will
> be T.30 only, as some peering partner changed the routes in the meantime.
>
> Best regards
> Peter
>
>
> On 02/23/17 20:38, Brian West wrote:
>
> What revision of FreeSWITCH are you on?  These behaviors have been
> solidified in 1.6.15, Sadly MOST of the details of how it all works is
> locked up in my head, I've just not had the time to write it all down and
> update the documentation.  Can you outline all the scenarios you need to
> cover and if you have the ability know ahead of time which one needs to
> apply?
>
> /b
>
>
> On Thu, Feb 23, 2017 at 8:23 AM, Peter Steinbach <lists at telefaks.de>
> wrote:
>
>> Hello,
>>
>> we have the following screnario
>>
>>    - The caller, a local fax machine (Fax, ATA and Freeswitch) with
>>    T.38, is behind a slow line so therefore T.38 is needed here
>>    - a central Freeswitch fax server serves as a central gateway to the
>>    SIP provider
>>    - the SIP provider sometimes reinvites T.38 and sometimes not.
>>    - In case, the SIP provider does NOT answer with T.38, the central
>>    Freeswitch fax server shall act as a T.38 gateway and transcode between
>>    local fax (T.38) and SIP provider (T.30)
>>    - In case, the SIP provider does answer with T.38, the central
>>    Freeswitch fax server shall pass T.38 to the local fax ,machine (T.38)
>>
>> How we set this up
>>
>>    - according to the wiki we have a dialplan as follows: (this is what
>>    CLI shows)
>>
>> Dialplan: sofia/internalvpn2/49274xxxxx at 10.7.0.1 Action
>> set(fax_enable_t38=true)
>> Dialplan: sofia/internalvpn2/49274xxxxx at 10.7.0.1 Action
>> set(fax_enable_t38_request=true)
>> Dialplan: sofia/internalvpn2/49274xxxxx at 10.7.0.1 Action
>> set(execute_on_answer=t38_gateway self)
>> Dialplan: sofia/internalvpn2/49274xxxxx at 10.7.0.1 Action ring_ready()
>> Dialplan: sofia/internalvpn2/49274xxxxx at 10.7.0.1 Action bridge(
>> sofia/gateway/sip.provider.com/xxxxxxxxxxxxxxx at sip.provider.com)
>>
>> In all profiles, we have
>> <param name="t38-passthru" value="true"/>
>>
>> In case, the SIP provider does NOT answer with T.38, it works. The
>> caller's side is reinvited with T.38, the SIP provider side is T.30
>>
>> What the problem is: When the SIP provider' side IS answering with T.38,
>> then (sometimes, when I send it via cli from the local freeswitch near the
>> sending fax machine, it works)
>>
>>    - caller initiates a call
>>    - call is bridged to the SIP provider' side
>>    - we receive a T.38 Reinvite from the SIP provider' side
>>    - Freeswitch sends Trying back to the the SIP provider' side
>>    - Freeswitch logs: set(execute_on_answer=t38_gateway self)
>>    - Freeswitch sends a T.38 Reinvite to the caller, establishes T.38 on
>>    this side.
>>    - But the T.38 Reinvite from the SIP provider' side is not answerered
>>    - So the SIP provider' sidehangs up after 10 sec of timeout.
>>
>> What are we doing wrong here? Is it possible to act as a t.38 passthrough
>> router and a T.38 gateway in the same dialplan?
>>
>>
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>> _________________________________________________________________________
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>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
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>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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