[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.
>

The canreinvite= option is an old setting, this is replaced by the 
directmedia= option in newer versions of Asterisk, it doesn't prevent a 
re-invite, it keeps the audio going through asterisk rather than 
negotiating an audio channel directly with the other endpoint.


The reason I suggested disabling udptl at that end is because my 
understanding of how the implementation of T.38 Gateway works on 
Asterisk is;

  1) it does not utilise any of the T.38 gateway features in spandsp

  2) the gateway will not step in if the originator negotiates T.38

Considering the other post you sent, are you suing IAX between the two 
Asterisk boxes?

To test the T.38 Gateway can work on your box in question set up an IAX 
modem and configure HylaFAX modem to use the iaxmodem on the box in 
question, test the gateway functionality.

When I tested Asterisk 11 a little while back I configured HylaFAX on my 
current system to communicate with an IAX modem on my Asterisk 11 test 
box and was able to observe the T.38 gateway function.

I can't tell from the information you've provided if the old Asterisk 
box is on the same network or having to traverse a WAN link to make the 
connection out through to your SIP provider.

Perhaps you could provide more information about your set up such as 
entries from your sip.conf, iax.conf, udptl.conf etc.


Larry.