I finally solved the issue by changing the resolution and the width of the TIFF file to one that is accepted by the fax standard. In my case I changed to a resolution of 96x96 and a width of 1728.<br><br>Now I am able to send faxes, but something weird is happening, the fax received in the fax-machine has the black and white colours inverted. Any ideas why this could be happening?<br>
<br>Best regards,<br><br>Santi<br><br><div class="gmail_quote">On Tue, Mar 10, 2009 at 6:53 PM, Santiago Gimeno <span dir="ltr"><<a href="mailto:santiago.gimeno@gmail.com">santiago.gimeno@gmail.com</a>></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;">Thanks for the tip. Sadly, it didn'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<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson <span dir="ltr"><<a href="mailto:creslin@digium.com" target="_blank">creslin@digium.com</a>></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>Santiago Gimeno wrote:<br>
> Hello,<br>
><br>
> Thanks everybody for the answers.<br>
><br>
> >Could be. Would you post the Cisco config relevant to this?<br>
><br>
> dial-peer voice 5 voip<br>
> description ** **<br>
> preference 1<br>
> destination-pattern 1…<br>
> voice-class codec 1<br>
> session protocol sipv2<br>
> session target ipv4:1.1.1.1<br>
> session transport udp<br>
> dtmf-relay rtp-nte<br>
> fax-relay ecm disable<br>
<br>
</div>I think, that at least if you'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><br>
> fax nsf 000000<br>
> fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through<br>
> g711alaw<br>
> no vad<br>
><br>
><br>
> >And upon further examination... don't put T38CALL in as a variable. It<br>
> will cause the initial INVITE to only<br>
> >have T38. Leave it out and things should hopefully reinvite.<br>
><br>
> I have removed the T38CALL variable and it looks better but it still<br>
> doesn't work.<br>
> Now asterisk sends an initial INVITE with audio media in the SDP. The<br>
> CISCO accepts this call after contacting the fax-machine. Then the CISCO<br>
> sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE.<br>
> But finally the fax transmission fails and the asterisk verbose trace is:<br>
><br>
> *CLI> -- Attempting call on SIP/080913216002@outbound-calls for<br>
> 22222@fax-out:1 (Retry 1)<br>
> == Using SIP RTP CoS mark 5<br>
> == Using UDPTL CoS mark 5<br>
> > Channel SIP/outbound-calls-0822aae8 was answered.<br>
> == Starting SIP/outbound-calls-0822aae8 at fax-out,22222,1 failed so<br>
> falling back to exten 's'<br>
> -- Executing [s@fax-out:1] Set("SIP/outbound-calls-0822aae8",<br>
> "FAXFILE=/root/santi/fax/prueba.tif") in new stack<br>
> -- Executing [s@fax-out:2]<br>
> SIPDtmfMode("SIP/outbound-calls-0822aae8", "inband") in new stack<br>
> -- Executing [s@fax-out:3] SendFAX("SIP/outbound-calls-0822aae8",<br>
> "/root/santi/fax/prueba.tif") in new stack<br>
> [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error<br>
> transmitting fax. result=11: Far end cannot receive at the resolution of<br>
> the image.<br>
> [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error<br>
> == Spawn extension (fax-out, s, 3) exited non-zero on<br>
> 'SIP/outbound-calls-0822aae8'<br>
><br>
> Any ideas?<br>
><br>
> Thanks. Best regards,<br>
><br>
> Santi<br>
><br>
><br>
><br>
> On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp <<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a><br>
</div></div><div>> <mailto:<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>>> wrote:<br>
> ><br>
> > ----- "Santiago Gimeno" <<a href="mailto:santiago.gimeno@gmail.com" target="_blank">santiago.gimeno@gmail.com</a><br>
</div><div>> <mailto:<a href="mailto:santiago.gimeno@gmail.com" target="_blank">santiago.gimeno@gmail.com</a>>> wrote:<br>
> ><br>
> > ><br>
> > > **The call-file I'm using is:<br>
> > ><br>
> > > Channel: SIP/080999999999@outbound-<br>
> > > calls<br>
> > > MaxRetries: 3<br>
> > > WaitTime: 30<br>
> > > Set: LOCALSTATIONID=22222<br>
> > > Set: LOCALHEADERINFO=T38 fax<br>
> > > Set: T38CALL=1<br>
> > > Set: T38TXDETECT=yes<br>
> > > CallerID: 22222<br>
> > > Context: fax-out<br>
> > > Extension: 22222<br>
> > > priority:1<br>
> > ><br>
> ><br>
> > And upon further examination... don't put T38CALL in as a variable.<br>
> It will cause the initial INVITE to only<br>
> > have T38. Leave it out and things should hopefully reinvite.<br>
> ><br>
> > --<br>
> > Joshua Colp<br>
> > Digium, Inc. | Software Developer<br>
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
</div>> > Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> <<a href="http://www.digium.com" target="_blank">http://www.digium.com</a>> &<br>
> <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a> <<a href="http://www.asterisk.org" target="_blank">http://www.asterisk.org</a>><br>
<div>> ><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>
><br>
><br>
</div>> ------------------------------------------------------------------------<br>
<div><div></div><div>><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>
<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>
</div></div></blockquote></div><br>