asterisk users - Jul 2012 - T.38 Gateway

Hi everyone,

we're trying to setup a fax gateway from PRI to SIP. Our test setup
consists of two asterisk 10 machines connected over PRI and a third one
which is acting as a fax endpoint over SIP.

On the machine which is receiving over PRI, we see the following message:

[Jul 23 12:54:23] DEBUG[18288] res_fax.c: detected v21 preamble on
SIP/xxx.xxx.xxx.xxx-00000001, but DAHDI/i1/-2 does not support T.38 for
T.38 gateway session

The relevant dialplan part looks like this:

[from-pstn]
exten => _X,1,Verbose(1,"Entering from-pstn context")
exten => _X,n,Set(FAXOPT(gateway)=yes)
exten => _X,n,Dial(SIP/12345 at xxx.xxx.xxx.xxx)

Fax capabilites are working apparently:

*CLI> fax show capabilities

Registered FAX Technology Modules:

Type            : Spandsp
Description     : Spandsp FAX Driver
Capabilities    : SEND RECEIVE T.38 G.711 GATEWAY

Any idea what could be wrong?

Thanks in advance for any help
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On 7/23/2012 6:30 AM, Paul Goldbaum wrote:
>
> Any idea what could be wrong?
>
> Thanks in advance for any help
>
Will probably help to list your Asterisk,DAHDI, SpanDSP software 
versions and pastebin a more comprehensive debug log covering what 
happens before and after the line you reference.

Thanks,

-- 
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org

On 07/23/2012 06:30 AM, Paul Goldbaum wrote:
> Hi everyone,
>
> we're trying to setup a fax gateway from PRI to SIP. Our test setup
> consists of two asterisk 10 machines connected over PRI and a third one
> which is acting as a fax endpoint over SIP.
This 'third' system is the one responsible for initiating the switchover
to T.38 mode for the call, since it is the last element in the path to 
the eventual called FAX terminal.
> On the machine which is receiving over PRI, we see the following message:
>
> [Jul 23 12:54:23] DEBUG[18288] res_fax.c: detected v21 preamble on
> SIP/xxx.xxx.xxx.xxx-00000001, but DAHDI/i1/-2 does not support T.38 for
> T.38 gateway session
This is expected behavior. If you run a gateway on a SIP channel, it 
will naturally assume that *that* channel is not capable of T.38 support 
(thus the need for a gateway) and it will attempt to request T.38 
support from the channel it is bridged to.

In your configuration, the Asterisk system acting as SIP FAX endpoint 
should *itself* request a switch to T.38 on the SIP channel before it 
ever emits any FAX tones at all.  The system in the middle will receive 
this T.38 negotiation request, and the gateway code (since it has been 
enabled) will realize that the DAHDI channel is not T.38 capable, and it 
will step in to provide T.38 gateway services.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org