[asterisk-users] T38 Passthrough Verification

[Andreas van dem Helge]

The call is still going to show up as the codec with which the voice
segment was established.

Have you viewed the SIP debug messages and confirmed that T.38 is not
being used?

FWIW the device that is receiving the T.38 fax (generally callee)
should be issuing the T.38 re-invite, so you might want to start at
that end.

Make sure t38pt_udptl = yes  is defined.


On Thu, May 8, 2008 at 8:55 AM, JR Richardson <jmr.richardson at gmail.com> wrote:
>> JR Richardson wrote:
>> > I have 1.4.9.1 setup, with the compiler flags enabled for T38, and
>> > have a Mediatrix 2102 and a Linksys SPA 8000-G1.  I can pass faxes
>> > between devices but can't seem to invoke T38 pt UDPTL.  It's enabled
>> > in sip.conf [general] and well as the [peer].
>> >
>> > I get an error at the CLI:
>> > WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite
>> > after T38 session not handled yet !
>> >
>> > sip show channels shows the call setup with ulaw.
>>
>> Try setting canreinvite=no for the peer doing T.38.  It looks like the
>> code in
>> Asterisk 1.4 will not allow re-invites for an established T.38 passthrough
>> call.
>
> I saw a post about the re-invites, so I tried it both ways,
> canreinvite=yes/no with the same results.
>
> Thanks.
>
> JR
> ---
> JR Richardson
> Engineering for the Masses
>
>
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