[asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?

[Larry Moore]

On 24/10/2014 12:49 AM, Tim Nelson wrote:
> ----- Original Message -----
>>
>>
>> On 23/10/2014 10:07 PM, Larry Moore wrote:
>>>
>>>
>>> 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.
>>>
>>
>> Have you had a look at
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
>>
>> As an exercise you could disable T.38 on 'Asterisk calling system',
>> if
>> you have an ATA which is originating the call to 'Asterisk calling
>> system' disable T.38 on that device too and disable in your sip.conf
>> using t38pt_udptl=no.
>>
>> If you are using SendFax() on 'Asterisk calling system' ensure T.38
>> is
>> not able to be used.
>>
>> If using an ATA connecting to 'Asterisk calling system' ensure you
>> have
>> set in your peer's configuration canreinvite=no or directmedia=no,
>> depending on the version of Asterisk you are running on this system.
>>
>> On Asterisk system in '(box in question)' set directmedia=no for the
>> peer which is connecting to 'SIP Provider' and also to 'Asterisk
>> calling
>> system', you may want to set setvar=FAXOPT(gateway)=yes in your peer
>> config to 'SIP Provider' otherwise it will need to be set in your
>> dialplan.
>>
>> Set your verbose&  debug to at least 3 on '(box in question)',
>> possibly
>> a little higher and send a fax - you may now see the Fax Gateway
>> detect
>> CED. Not sure if this is suppressed in
>>
>> You may want enable udptl debugging on '(box in question)'.
>>
>
> I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP.
>
> Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly.
>
> --Tim
>

It would seem for Asterisk 11 and T.38 Gateway work for an IAX channel 
you require the following;

IAX2 -> SIP -> T.38 Gateway -> ITSP (SIP)

Where as it would be nicer if it would accept acting as a gateway for an 
IAX channel i.e.;

IAX2 - T.38 Gateway -> ITSP (SIP)

If an IAX2 channel is connected directly to a context with 
FAXOPT(t38gateway) enabled I see 'ast_rtp_read: RTP Read too short' 
messages and a failed transmission, the same is observed if using SIP 
with udptl=no instead of IAX2 channel;

SIP (udptl=no) -> T.38 Gateway -> ITSP (SIP).

Not sure if this is by design!

Maybe time for another friendly chat with your ITSP in the hope they can 
resolve the issue.

Larry.