[asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102

[Atis Lezdins]

On Thu, Aug 14, 2008 at 4:46 AM, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and
> Linksys 2102:
>
> PRI> <TNT> <SIP> <Asterisk> <2102> <SharpFax
>
> Faxing either direction, the call sets up with ulaw rtp, when fax tones hit
> the line, both the TNT and the 2102 switch to t38 and udptl packets fly
> through Asterisk.  All looks good, but, once udptl sets up, every few
> seconds, I get a warning: 'rtp Read too short' on the Asterisk CLI from the
> TNT side of the session.  Faxes never complete, not even a half page,
> nothing, transmission just ends.
>
> There are only a few parameters on the TNT that effect t38 and I've adjusted
> them all with no change in the results.
>
> Pretty much the same results when testing t38 pass through to a Cisco pri
> gateway as well.
>
> So my question is: Does anyone else have this solution working and wouldn't
> not mind sharing configs?

Hi,

I have T38 passtrough working in following configuration:

Callweaver -> Asterisk -> SIP proxy (provider) -> provider's switch.

I never had received such errors, i suspect it must be problem of one
of those devices (sending bad packets), so perhaps you can check them
by skipping Asterisk, and also try connecting another different device
(i.e. Callweaver) to any of endpoints.

Alternatively you can try to get Asterisk out of UDPTL path, by
enabling re-invites.

As for configuration, i had added only t38pt_udptl=yes in [general]
section and peer section and everything worked.

my sip.conf:

[general]
port = 5060
bindaddr = 0.0.0.0
context = default-external
tos_sip=0x18
tos_audio=0x18
callerid = Unknown
dtmfmode=rfc2833
registertimeout=200
ignoreregexpire=no
limitonpeer=yes
notifyringing=no
notifyhold=no
allowsubscribe=yes
rtcachefriends=yes
t38pt_udptl=yes


[callweaver]
type=friend
host=127.0.0.1
permit=127.0.0.1
context=callweaver_out
port=7060
allow=all
canreinvite=no
t38pt_udptl=yes

; note - SIP provider don't have entry, it's dialed by IP.


Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835