asterisk users - Apr 2016 - T.38 with Audiocodes gateway

Hello,

I'm helping a colleague (*) which has the following setup:

ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 ---  <PJSIP>--
Audiocodes MP-112 ---  <FXO/FXS> --- Fax machine

My issue is the following :
Audiocodes gateway reject INVITEs with 488 Not Acceptable Here

It seems this gateway requires t38 settings to be present in SDP body in
the very first INVITE.

My questions are the following:

1. I expected T.38 to exclusively work with reINVITE where calls are
established as normal voice calls (PCMA/PCMU in SDP, for instance) and then
upgraded to T.38 (when CNG is detected, for instance).
Have you ever heard of T.38 sessions being established right from the start
(ie with T.38 settings in the first INVITE) ?

2. Is it possible to configure Asterisk to pass T.38 settings in SDP in the
first INVITE it sends ?

3. Any suggestion with Audiocodes gateway ?


Regards

(*) That's the reason why I don't have as many details as I would like,
at
the moment.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160429/4e630a03/attachment.html>

On Fri, Apr 29, 2016 at 1:34 AM, Olivier <oza.4h07 at gmail.com>
wrote:
> Hello,
>
> I'm helping a colleague (*) which has the following setup:
>
> ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- 
<PJSIP>--
> Audiocodes MP-112 ---  <FXO/FXS> --- Fax machine
>
> My issue is the following :
> Audiocodes gateway reject INVITEs with 488 Not Acceptable Here
>
> It seems this gateway requires t38 settings to be present in SDP body in
the
> very first INVITE.
>
> My questions are the following:
>
> 1. I expected T.38 to exclusively work with reINVITE where calls are
> established as normal voice calls (PCMA/PCMU in SDP, for instance) and then
> upgraded to T.38 (when CNG is detected, for instance).
> Have you ever heard of T.38 sessions being established right from the start
> (ie with T.38 settings in the first INVITE) ?
No.  It would seem to be extremely broken if it denies a call based on
a lack of T.38 sdp parameters on the initial INVITE.
> 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in the
> first INVITE it sends ?
>
> 3. Any suggestion with Audiocodes gateway ?
Look for T.38 settings maybe?  See if there is something keeping you
from sending an initial invite with non-T.38 SDP....?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA