Thanks for the tip. Sadly, it didn&#39;t work. I keep getting the same error:<br><br>[Mar 10 18:49:48] WARNING[18855]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image.<br>
<br>regards,<br><br>Santi<br><br><div class="gmail_quote">On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson <span dir="ltr">&lt;<a href="mailto:creslin@digium.com">creslin@digium.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">Santiago Gimeno wrote:<br>
&gt; Hello,<br>
&gt;<br>
&gt; Thanks everybody for the answers.<br>
&gt;<br>
&gt;  &gt;Could be. Would you post the Cisco config relevant to this?<br>
&gt;<br>
&gt; dial-peer voice 5 voip<br>
&gt; description ** **<br>
&gt; preference 1<br>
&gt; destination-pattern 1…<br>
&gt; voice-class codec 1<br>
&gt; session protocol sipv2<br>
&gt; session target ipv4:1.1.1.1<br>
&gt; session transport udp<br>
&gt; dtmf-relay rtp-nte<br>
&gt; fax-relay ecm disable<br>
<br>
</div>I think, that at least if you&#39;re using T.38, you may want to try<br>
enabling ECM.  ECM can cause significant problems in a high-packet loss,<br>
non-T.38 environment, but I would think that in a T.38 environment, if<br>
you can keep ECM enabled, that would be a good thing.<br>
<br>
Matthew Fredrickson<br>
Digium, Inc.<br>
<div><div></div><div class="h5"><br>
&gt; fax nsf 000000<br>
&gt; fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through<br>
&gt; g711alaw<br>
&gt; no vad<br>
&gt;<br>
&gt;<br>
&gt;  &gt;And upon further examination... don&#39;t put T38CALL in as a variable. It<br>
&gt; will cause the initial INVITE to only<br>
&gt;  &gt;have T38. Leave it out and things should hopefully reinvite.<br>
&gt;<br>
&gt; I have removed the T38CALL variable and it looks better but it still<br>
&gt; doesn&#39;t work.<br>
&gt; Now asterisk sends an initial INVITE with audio media in the SDP. The<br>
&gt; CISCO accepts this call after contacting the fax-machine. Then the CISCO<br>
&gt; sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE.<br>
&gt; But finally the fax transmission fails and the asterisk verbose trace is:<br>
&gt;<br>
&gt; *CLI&gt;     -- Attempting call on SIP/080913216002@outbound-calls for<br>
&gt; 22222@fax-out:1 (Retry 1)<br>
&gt;   == Using SIP RTP CoS mark 5<br>
&gt;   == Using UDPTL CoS mark 5<br>
&gt;        &gt; Channel SIP/outbound-calls-0822aae8 was answered.<br>
&gt;   == Starting SIP/outbound-calls-0822aae8 at fax-out,22222,1 failed so<br>
&gt; falling back to exten &#39;s&#39;<br>
&gt;     -- Executing [s@fax-out:1] Set(&quot;SIP/outbound-calls-0822aae8&quot;,<br>
&gt; &quot;FAXFILE=/root/santi/fax/prueba.tif&quot;) in new stack<br>
&gt;     -- Executing [s@fax-out:2]<br>
&gt; SIPDtmfMode(&quot;SIP/outbound-calls-0822aae8&quot;, &quot;inband&quot;) in new stack<br>
&gt;     -- Executing [s@fax-out:3] SendFAX(&quot;SIP/outbound-calls-0822aae8&quot;,<br>
&gt; &quot;/root/santi/fax/prueba.tif&quot;) in new stack<br>
&gt; [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error<br>
&gt; transmitting fax. result=11: Far end cannot receive at the resolution of<br>
&gt; the image.<br>
&gt; [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error<br>
&gt;   == Spawn extension (fax-out, s, 3) exited non-zero on<br>
&gt; &#39;SIP/outbound-calls-0822aae8&#39;<br>
&gt;<br>
&gt; Any ideas?<br>
&gt;<br>
&gt; Thanks. Best regards,<br>
&gt;<br>
&gt; Santi<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp &lt;<a href="mailto:jcolp@digium.com">jcolp@digium.com</a><br>
</div></div><div class="im">&gt; &lt;mailto:<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>&gt;&gt; wrote:<br>
&gt;  &gt;<br>
&gt;  &gt; ----- &quot;Santiago Gimeno&quot; &lt;<a href="mailto:santiago.gimeno@gmail.com">santiago.gimeno@gmail.com</a><br>
</div><div class="im">&gt; &lt;mailto:<a href="mailto:santiago.gimeno@gmail.com">santiago.gimeno@gmail.com</a>&gt;&gt; wrote:<br>
&gt;  &gt;<br>
&gt;  &gt; &gt;<br>
&gt;  &gt; &gt; **The call-file I&#39;m using is:<br>
&gt;  &gt; &gt;<br>
&gt;  &gt; &gt; Channel: SIP/080999999999@outbound-<br>
&gt;  &gt; &gt; calls<br>
&gt;  &gt; &gt; MaxRetries: 3<br>
&gt;  &gt; &gt; WaitTime: 30<br>
&gt;  &gt; &gt; Set: LOCALSTATIONID=22222<br>
&gt;  &gt; &gt; Set: LOCALHEADERINFO=T38 fax<br>
&gt;  &gt; &gt; Set: T38CALL=1<br>
&gt;  &gt; &gt; Set: T38TXDETECT=yes<br>
&gt;  &gt; &gt; CallerID: 22222<br>
&gt;  &gt; &gt; Context: fax-out<br>
&gt;  &gt; &gt; Extension: 22222<br>
&gt;  &gt; &gt; priority:1<br>
&gt;  &gt; &gt;<br>
&gt;  &gt;<br>
&gt;  &gt; And upon further examination... don&#39;t put T38CALL in as a variable.<br>
&gt; It will cause the initial INVITE to only<br>
&gt;  &gt; have T38. Leave it out and things should hopefully reinvite.<br>
&gt;  &gt;<br>
&gt;  &gt; --<br>
&gt;  &gt; Joshua Colp<br>
&gt;  &gt; Digium, Inc. | Software Developer<br>
&gt;  &gt; 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
</div>&gt;  &gt; Check us out at:  <a href="http://www.digium.com" target="_blank">www.digium.com</a> &lt;<a href="http://www.digium.com" target="_blank">http://www.digium.com</a>&gt;  &amp;<br>
&gt; <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a> &lt;<a href="http://www.asterisk.org" target="_blank">http://www.asterisk.org</a>&gt;<br>
<div class="im">&gt;  &gt;<br>
&gt;  &gt; _______________________________________________<br>
&gt;  &gt; -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
&gt;  &gt;<br>
&gt;  &gt; asterisk-users mailing list<br>
&gt;  &gt; To UNSUBSCRIBE or update options visit:<br>
&gt;  &gt;   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
&gt;<br>
&gt;<br>
</div>&gt; ------------------------------------------------------------------------<br>
<div><div></div><div class="h5">&gt;<br>
&gt; _______________________________________________<br>
&gt; -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
&gt;<br>
&gt; asterisk-users mailing list<br>
&gt; To UNSUBSCRIBE or update options visit:<br>
&gt;    <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>