Thanks for the tip. Sadly, it didn't work. I keep getting the same error: [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.
regards, Santi On Tue, Mar 10, 2009 at 6:36 PM, Matthew Fredrickson <cres...@digium.com>wrote: > Santiago Gimeno wrote: > > Hello, > > > > Thanks everybody for the answers. > > > > >Could be. Would you post the Cisco config relevant to this? > > > > dial-peer voice 5 voip > > description ** ** > > preference 1 > > destination-pattern 1… > > voice-class codec 1 > > session protocol sipv2 > > session target ipv4:1.1.1.1 > > session transport udp > > dtmf-relay rtp-nte > > fax-relay ecm disable > > I think, that at least if you're using T.38, you may want to try > enabling ECM. ECM can cause significant problems in a high-packet loss, > non-T.38 environment, but I would think that in a T.38 environment, if > you can keep ECM enabled, that would be a good thing. > > Matthew Fredrickson > Digium, Inc. > > > fax nsf 000000 > > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through > > g711alaw > > no vad > > > > > > >And upon further examination... don't put T38CALL in as a variable. It > > will cause the initial INVITE to only > > >have T38. Leave it out and things should hopefully reinvite. > > > > I have removed the T38CALL variable and it looks better but it still > > doesn't work. > > Now asterisk sends an initial INVITE with audio media in the SDP. The > > CISCO accepts this call after contacting the fax-machine. Then the CISCO > > sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE. > > But finally the fax transmission fails and the asterisk verbose trace is: > > > > *CLI> -- Attempting call on SIP/080913216...@outbound-calls for > > 22...@fax-out:1 (Retry 1) > > == Using SIP RTP CoS mark 5 > > == Using UDPTL CoS mark 5 > > > Channel SIP/outbound-calls-0822aae8 was answered. > > == Starting SIP/outbound-calls-0822aae8 at fax-out,22222,1 failed so > > falling back to exten 's' > > -- Executing [...@fax-out:1] Set("SIP/outbound-calls-0822aae8", > > "FAXFILE=/root/santi/fax/prueba.tif") in new stack > > -- Executing [...@fax-out:2] > > SIPDtmfMode("SIP/outbound-calls-0822aae8", "inband") in new stack > > -- Executing [...@fax-out:3] SendFAX("SIP/outbound-calls-0822aae8", > > "/root/santi/fax/prueba.tif") in new stack > > [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error > > transmitting fax. result=11: Far end cannot receive at the resolution of > > the image. > > [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission > error > > == Spawn extension (fax-out, s, 3) exited non-zero on > > 'SIP/outbound-calls-0822aae8' > > > > Any ideas? > > > > Thanks. Best regards, > > > > Santi > > > > > > > > On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp <jc...@digium.com > > <mailto:jc...@digium.com>> wrote: > > > > > > ----- "Santiago Gimeno" <santiago.gim...@gmail.com > > <mailto:santiago.gim...@gmail.com>> wrote: > > > > > > > > > > > **The call-file I'm using is: > > > > > > > > Channel: SIP/080999999...@outbound- > > > > calls > > > > MaxRetries: 3 > > > > WaitTime: 30 > > > > Set: LOCALSTATIONID=22222 > > > > Set: LOCALHEADERINFO=T38 fax > > > > Set: T38CALL=1 > > > > Set: T38TXDETECT=yes > > > > CallerID: 22222 > > > > Context: fax-out > > > > Extension: 22222 > > > > priority:1 > > > > > > > > > > And upon further examination... don't put T38CALL in as a variable. > > It will cause the initial INVITE to only > > > have T38. Leave it out and things should hopefully reinvite. > > > > > > -- > > > Joshua Colp > > > Digium, Inc. | Software Developer > > > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > > > Check us out at: www.digium.com <http://www.digium.com> & > > www.asterisk.org <http://www.asterisk.org> > > > > > > _______________________________________________ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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