Ricardo,<br>Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf?<br><br>Thanks,<br><br><b><i>Ricardo Carvalho &lt;rcarvalho@iric.up.pt&gt;</i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> No, T.38 doesn't work with Asterisk. Only works with Asterisk <br>t38passthrough patch that you can find at URL: <br>http://bugs.digium.com/file_download.php?file_id=9335&amp;type=bug<br>For me it only worked well with patch for version 1.2.4 of Asterisk.<br><br>Regards,<br><br>Ricardo.<br><br><br><br><br><br><br>Kokfoo Soo wrote:<br>&gt; Is T.38 fax work through Asterisk? I have the config below in my <br>&gt; sip.conf, but the fax doesn't work and give me the CLI lines below. My <br>&gt; current version is 1.2.10. Please help.<br>&gt;<br>&gt; [Inboundtopbx]<br>&gt; type=friend<br>&gt; context=pbx<br>&gt; host=10.18.188.84<br>&gt; insecure=port<br>&gt;
 dtmfmode=rfc2833<br>&gt; canreinvite=no<br>&gt; disallow=all<br>&gt; allow=g729<br>&gt; allow=ulaw<br>&gt; t38pt_udptl=yes<br>&gt; t38pt_rtp=no<br>&gt; t38pt_tcp=no<br>&gt;<br>&gt; [OutboundfromPBX]<br>&gt; type=peer<br>&gt; host=10.18.161.222              <br>&gt; canreinvite=no<br>&gt; dtmfmode=rfc2833<br>&gt; disallow=all<br>&gt; allow=g729<br>&gt; qualify=yes<br>&gt; t38pt_udptl=yes<br>&gt; t38pt_rtp=no<br>&gt; t38pt_tcp=no<br>&gt;<br>&gt; &lt;-- SIP read from 10.18.188.84:50096:<br>&gt; ACK sip:17815057304@10.18.161.237:5060 SIP/2.0<br>&gt; Via: SIP/2.0/UDP  10.18.188.84:5060<br>&gt; From: <sip:7350@10.18.188.84>;tag=19D429E8-2084<br>&gt; To: <sip:17815057304@10.18.161.237>;tag=as3c87a22e<br>&gt; Date: Tue, 05 Sep 2006 19:42:28 GMT<br>&gt; Call-ID: 7F23A1F9-3C4D11DB-A303B82B-9F58A83F@10.18.188.84<br>&gt; Max-Forwards: 6<br>&gt; Content-Length: 0<br>&gt; CSeq: 101 ACK<br>&gt;<br>&gt;<br>&gt; --- (9 headers 0 lines)---<br>&gt; Sep  5 15:30:31 NOTICE[25233]: rtp.c:564
 ast_rtp_read: Unknown RTP <br>&gt; codec 100 received<br>&gt; Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP <br>&gt; codec 100 received<br>&gt; Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP <br>&gt; codec 100 received<br>&gt; Sep  5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP <br>&gt; codec 100 received<br>&gt; Sep  5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown <br>&gt; SDP media type in offer: image 16406 udptl t38<br>&gt;<br>&gt; ------------------------------------------------------------------------<br>&gt; Yahoo! Messenger with Voice. Make PC-to-Phone Calls <br>&gt; <http: //us.rd.yahoo.com/mail_us/taglines/postman1/*http://us.rd.yahoo.com/evt="39663/*http://voice.yahoo.com"> <br>&gt; to the US (and 30+ countries) for 2¢/min or less.<br>&gt; ------------------------------------------------------------------------<br>&gt;<br>&gt; _______________________________________________<br>&gt;
 --Bandwidth and Colocation provided by Easynews.com --<br>&gt;<br>&gt; asterisk-users mailing list<br>&gt; To UNSUBSCRIBE or update options visit:<br>&gt;    http://lists.digium.com/mailman/listinfo/asterisk-users<br>&gt;   <br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>   http://lists.digium.com/mailman/listinfo/asterisk-users<br></http:></sip:17815057304@10.18.161.237></sip:7350@10.18.188.84></blockquote><br><p>&#32;
                <hr size=1>How low will we go? Check out Yahoo! Messenger’s low <a href="http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com"> PC-to-Phone call rates.