[asterisk-users] Fax with T.38

Sat Feb 24 13:30:38 MST 2007

Hi Bill,

I'm in exactly the same boat with T.38 and OpenPBX.  I too think the Cisco-T.38 Gateway is the most practical at this moment.  Where are you on testing this and can you share the 3660 config?  In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems.

Jon

-----Original message-----
From: "Bill Gibbs" bgibbs at edurotech.com
Date: Thu, 22 Feb 2007 15:02:18 -0500
To: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users at lists.digium.com
Subject: RE: [asterisk-users] Fax with T.38

> Ray,
> 
> I have been playing with OpenPBX.  My core servers are Asterisk so I was playing around with their T38Gateway application.  Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail.  Clearly T38 was working though, debug output was full of T38 talk.  However the wiki clearly states it's experimental still.
> 
> I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP.  I am going to set that up to talk to * 1.4.0 and do T38 pass through.  I to will be doing NAT for the ATAs so...hopefully it will work.  We shall see.
> 
> So my call flow will be
> 
> PRI -> Asterisk 1.2.x
> Out the 2nd PRI to the 3660
> 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA.
> 
> I have the 3660 there to take the call via TDM and convert to T38.  I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned.
> 
> That's how I plan to set it up.
> 
> Bill
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ray Jackson
> Sent: Wednesday, February 21, 2007 10:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Fax with T.38
> 
> 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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