<div>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.</div>
<div> </div>
<div><br> </div>
<div><span class="gmail_quote">On 2/19/07, <b class="gmail_sendername">Thomas Deillon</b> <<a href="mailto:Thomas.Deillon@smart-telecom.ch">Thomas.Deillon@smart-telecom.ch</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi all,<br><br>I make others tests.<br>Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog Fax2
<br><br>It works only if I use canreinvite= yes.<br>But all my clients are behind a Nat without ALG or stun stuffs...<br><br>Do you know if canreinvite= yes it's the only way to make it works??<br><br>Thanks a lot for your help,
<br><br>Thomas<br><br><br><br>-----Message d'origine-----<br>De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">
asterisk-users-bounces@lists.digium.com</a>] De la part de Thomas Deillon<br>Envoyé: jeudi, 15. février 2007 11:26<br>À: Asterisk Users Mailing List - Non-Commercial Discussion<br>Objet: [asterisk-users] Fax with T.38<br>
<br>Hi all,<br><br>I make mistakes in my explanation, so I will try to re-explain my problem…<br><br>I want to send fax with FoIP.<br>Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2
<br><br>In the Patton SN4960 configuration I have :<br>profile voip FOIP<br>codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression<br>codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression<br>codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression
<br>dtmf-relay signaling<br>dejitter-max-delay 100<br>fax transmission 1 relay t38-udp<br>fax redundancy low-speed 2 high-speed 1<br>fax detection fax-frames<br>modem transmission 1 bypass g711alaw64k<br>modem bypass-method nse
<br><br>On Patton M-ATA :<br>1. codec alaw<br>2. codec ulaw<br>3. codec g729<br>No silence suppression on these codecs.<br>I not use this option "FAX without T.38(Use G.711 fax)"<br><br><br>On asterisk side I have:<br>[general]
<br>context=default<br>bindport=5060<br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a> <br>srvlookup=yes<br>disallow=all<br>allow=alaw<br>dtmfmode = rfc2833<br>rtcachefriends=yes<br>realm=vtxvoip<br>useragent=VTX SIP<br>rtupdate=yes
<br>language=en<br>tos=184<br>notifyringing=yes<br>t38pt_udptl=yes<br><br>And t38pt_udptl=yes in the 2 PATTONs sip accounts …<br><br><br>Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 ….<br>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:<br><br>[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)
<br>[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729<br>[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729
<br><br><br>What I really not understand it's why asterisk try to translate from ulaw to g729 !!!<br>I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file …<br><br>Do you have an idea for me ??
<br><br>Thanks a lot,<br><br>Thomas<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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