On 23/10/2014 3:55 AM, Tim Nelson wrote:
----- 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. :)


No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now.

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