[asterisk-users] Fax with T.38

[Ray Jackson]

Could anybody give me an authoritative answer on whether Asterisk can 
support T.38 pass-through when the clients are behind NAT?  We have 
Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) 
and would love to get T.38 going but have had no luck so far.  The 
following case:

http://bugs.digium.com/view.php?id=7844

...suggests that T.38 *does* now work for clients behind NAT but I have 
the latest SVN trunk but still cannot get it to work?  On the other side 
I have seen on this list only 2 weeks or so ago:

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html

This suggests that T.38 does *NOT* work behind NAT?  So, can anybody 
save me the trouble and tell me how it is.  Am I on a hiding to nothing 
trying to get T.38 going with NAT?  Please put me out of my misery! :)

Cheers,
Ray

PS. Does anybody know whether OpenPBX would support T.38 and NAT 
configurations?  This was my backup plan if I couldn't get it to go in 
Asterisk.

Thomas Deillon wrote:
> Yes, the canreinvite means Re invite, but there is a consequence in 
> Asterisk configuration.
> 
> For sure, all the signalisation traffic will go through the asterisk … 
> but for the RTP traffic?
> 
> If canreinvite = No, all RTP traffic will go through the Asterisk 
> (useful for NATed phoned without ALG/STUN/…)
> 
> If canreinvite = Yes, the phones will try to exchange RTP packets directly.
> 
>  
> 
> Do you thing there is a way to allow Re Invite (because you’re right) 
> without the RTP consequence?
> 
>  
> 
> Thanks a lot for your help,
> 
>  
> 
> Thomas
> 
>  
> 
> ------------------------------------------------------------------------
> 
> *De :* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *De la part de* Rajnish 
> Jain
> *Envoyé :* lundi, 19. février 2007 16:25
> *À :* Asterisk Users Mailing List - Non-Commercial Discussion
> *Objet :* Re: [asterisk-users] Fax with T.38
> 
>  
> 
> A T.38 fax call typically begins as a normal voice media call. The 
> call then dynamically switches over T.38 image media on detection of fax 
> handshake tones.  The dynamic modification of session from audio to 
> image is accomplished through SIP RE-INVITE messages. I would imagine 
> canreinvite= flag controls if an end-point is allowed to send/recv 
> RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 
> to work.
> 
>  
> 
> 
>  
> 
> On 2/19/07, *Thomas Deillon* <Thomas.Deillon at smart-telecom.ch 
> <mailto:Thomas.Deillon at smart-telecom.ch>> wrote:
> 
> Hi all,
> 
> I make others tests.
> Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog Fax2
> 
> It works only if I use canreinvite= yes.
> But all my clients are behind a Nat without ALG or stun stuffs...
> 
> Do you know if canreinvite= yes it's the only way to make it works??
> 
> Thanks a lot for your help,
> 
> Thomas
> 
> 
> 
> -----Message d'origine-----
> De: asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com> [mailto: 
> asterisk-users-bounces at lists.digium.com 
> <mailto:asterisk-users-bounces at lists.digium.com>] De la part de Thomas 
> Deillon
> Envoyé: jeudi, 15. février 2007 11:26
> À: Asterisk Users Mailing List - Non-Commercial Discussion
> Objet: [asterisk-users] Fax with T.38
> 
> Hi all,
> 
> I make mistakes in my explanation, so I will try to re-explain my problem…
> 
> I want to send fax with FoIP.
> Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA 
> ←Analog→ Analog Fax 2
> 
> In the Patton SN4960 configuration I have :
> profile voip FOIP
> codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression
> codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression
> codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression
> dtmf-relay signaling
> dejitter-max-delay 100
> fax transmission 1 relay t38-udp
> fax redundancy low-speed 2 high-speed 1
> fax detection fax-frames
> modem transmission 1 bypass g711alaw64k
> modem bypass-method nse
> 
> On Patton M-ATA :
> 1. codec alaw
> 2. codec ulaw
> 3. codec g729
> No silence suppression on these codecs.
> I not use this option "FAX without T.38(Use G.711 fax)"
> 
> 
> On asterisk side I have:
> [general]
> context=default
> bindport=5060
> bindaddr=0.0.0.0 <http://0.0.0.0>
> srvlookup=yes
> disallow=all
> allow=alaw
> dtmfmode = rfc2833
> rtcachefriends=yes
> realm=vtxvoip
> useragent=VTX SIP
> rtupdate=yes
> language=en
> tos=184
> notifyringing=yes
> t38pt_udptl=yes
> 
> And t38pt_udptl=yes in the 2 PATTONs sip accounts …
> 
> 
> Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 ….
> I received T.38 packets from the Patton sn4960 but no T.38 packets go 
> through the Asterisk …. And on the asterisk I have 3 WARNINGS:
> 
> [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 
> ast_channel_make_compatible: No path to translate from 
> SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8)
> [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to 
> find a codec translation path from alaw to g729
> [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to 
> find a codec translation path from alaw to g729
> 
> 
> What I really not understand it's why asterisk try to translate from 
> ulaw to g729 !!!
> I disallow all and allow just the alaw codec … more than this, I remove 
> the g729 licence file …
> 
> Do you have an idea for me ??
> 
> Thanks a lot,
> 
> Thomas
> 
> 
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