asterisk users - Oct 2014 - res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

Greetings- 

Working with the T.38 gateway functionality that is sparsely documented [1] ,
I'm attempting to get the following functional:

Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in
question) -> SIP Provider

The problem is: 

-The provider is not initiating a reinvite to T.38, even though it is 100%
supported
-Asterisk is not detecting the CNG tones from the far side of the call and
initiating a T.38 session on that call leg (with the SIP provider), but *DOES*
attempt to initiate a T.38 session with the calling Asterisk system (which
rejects with SIP/488 as expected)

So, how does one force a reinvite to T.38 on the outbound call leg in this
scenario? I did find the same problem from another user on the list in the
archives, but didn't find a solution contained within the responses [2] .

Thank you, 

--Tim 

[1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway 
[2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html 
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----- Original Message ----- 
> Greetings-
> Working with the T.38 gateway functionality that is sparsely
> documented [1], I'm attempting to get the following functional:
> Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
> in question) -> SIP Provider
> The problem is:
> -The provider is not initiating a reinvite to T.38, even though it is
> 100% supported
> -Asterisk is not detecting the CNG tones from the far side of the
> call and initiating a T.38 session on that call leg (with the SIP
> provider), but *DOES* attempt to initiate a T.38 session with the
> calling Asterisk system (which rejects with SIP/488 as expected)
> So, how does one force a reinvite to T.38 on the outbound call leg in
> this scenario? I did find the same problem from another user on the
> list in the archives, but didn't find a solution contained within
> the responses [2].
> Thank you,
> --Tim
> [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
> [2]
> http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html

*bump*

Any thoughts? I'm quite familiar with the T.38 functionality within
Callweaver, and a function is provided there to do exactly what I need (
SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and
better T.38 features such as "gateway" do not function, I am pressed
to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since
spandsp.org is down) for this job. :)

Thanks!

--Tim

On 22/10/2014 11:23 AM, Tim Nelson wrote:
> Greetings-
>
> Working with the T.38 gateway functionality that is sparsely documented
> [1], I'm attempting to get the following functional:
>
What type of endpoint are you using which is originating the call and is 
it T.38 capable?

Larry.