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" [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: "Asterisk Users Mailing List - Non-Commercial Discussion"[email protected] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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/[email protected]/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 :* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *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* <[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>> 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: [EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]> [mailto: > > [EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>] 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 > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
