Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Larry Moore omninet.net.au> writes: > > sip.conf > > [general] > faxdetect=t38 > > [sipcall.ch] > directmedia=no > > In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a > T.38 re-invite this will trigger the switch to the Fax extension. > > If this proves successful you can work on removing the Wait() from your > dialplan as Asterisk will remain in the audio path and should > successfully switch to the fax extension if extension 200 or 201 answer > a call that happens to be a fax. > > Larry. > Hi to all Sorry to bump this old thread. Well, I found a while ago finally the reason why T.38 doesn't work in conjunction with Swiss VoIP provider sipcall. Despite T.38 is stated as "supported", that provider does NOT support T.38. Their T.38 gateway has some fundamental negotiation problems, - it "exceeds the T4 timer of the T.30 protocol". Therefore, T.38 faxing does not work. http://wiki.innovaphone.com/index.php?title=Howto:Sipcall_business_-_SIP_Provider_Compatibility_Test Sipcall has confirmed me that they work now on a solution. Will see... Kind regards, Clemens -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote: Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) The T.38 connection will be attempted when ReceiveFax is called. The port number to use should be in the SDP information, yes, allow udp ports 4000-4999 in and out. If your firewall can be so configured you could set it to allow traffic in and out based upon the user ID Asterisk is running as, assuming it is using a unique unprivileged id. You may like to try the following to see if your SIP provider will initiate a T.38 re-invite. sip.conf [general] faxdetect=t38 [sipcall.ch] directmedia=no In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a T.38 re-invite this will trigger the switch to the Fax extension. If this proves successful you can work on removing the Wait() from your dialplan as Asterisk will remain in the audio path and should successfully switch to the fax extension if extension 200 or 201 answer a call that happens to be a fax. Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob may do i have to open more ports then udp 1:2 (RTP), udp 4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. after that i started udptl debug as well and now i'm getting lots of UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 152, len 11) and in between [Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short and in the end [Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling destruction for 1 ms [Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' failure, reason: 'fax session timed-out' (TIMEOUT) == Spawn extension (fax-rx, receive, 11) exited non-zero on 'SIP/sipcall.ch-0007' Thx, Jakob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data In udptl.conf set use_even_ports=yes and then issue a reload. You can confirm the settings have been applied by performing udptl show config. Change the the t38 line to read as; t38pt_udptl=yes,redundancy,maxdatagram=400 Reload sip and test. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed to initialize UDPTL, declining image stream [Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax will be received as an audio call rather than T.38, setting t38pt_udptl=no has turned off T.38. Do you know if your upstream provider supports T.38? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Thanks Am 03.02.2014 11:57, schrieb Larry Moore: On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 add directmedia=no setvar=FAXOPT(gateway)=no change insecure=port,invite [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) Do you want to keep your received faxes or is it OK to overwrite them the next time asterisk is re-started!? udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no You may want to change use_even_ports=yes You will need to restart Asterisk for this change. Some other suggestion if the above doesn't help are; faxdetect=cng t38pt_udptl=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 add directmedia=no setvar=FAXOPT(gateway)=no change insecure=port,invite [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) Do you want to keep your received faxes or is it OK to overwrite them the next time asterisk is re-started!? udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no You may want to change use_even_ports=yes You will need to restart Asterisk for this change. Some other suggestion if the above doesn't help are; faxdetect=cng t38pt_udptl=no Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes,redundancy,maxdatagram=400 directrtpsetup=yes disallow=all allow=ulaw allow=alaw and the corresponding Peer [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 the Dialplan [inbound] exten => _X.,1,Answer() exten => _X.,n,Set(DB(lastcaller/number)=${CALLERID(num)}) exten => _X.,n,GotoIf(${BLACKLIST()}?black,1) exten => _X.,n,Wait(2) exten => _X.,n,Dial(SIP/200&SIP/201,60,tToxX) exten => _X.,n,Goto(ausser-zeit,_X.,3) exten => _X.,n,Hangup() exten => fax,1,NoOp( FAX DETECTED ) exten => fax,n,Goto(fax-rx,receive,1) [fax-rx] exten => receive,1,NoOp( FAX RECEIVE ) exten => receive,n,Set(GLOBAL(FAXCOUNT)=$["${GLOBAL(FAXCOUNT)}" + "1"]) exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)}) exten => receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif) exten => receive,n,Set(GLOBAL(LASTFAXCALLERoNUM)=${CALLERID(num)}) exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)}) exten => receive,n,NoOp( SETTING FAXOPT ) exten => receive,n,Set(FAXOPT(ecm)=yes) exten => receive,n,Set(FAXOPT(headerinfo)=MYFAX RX) exten => receive,n,Set(FAXOPT(localstationid)=1234567890) exten => receive,n,Set(FAXOPT(maxrate)=14400) exten => receive,n,Set(FAXOPT(minrate)=2400) exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten => receive,n,NoOp( RECEIVING FAX : ${FAXFILE} ) exten => receive,n,ReceiveFAX(/var/spool/asterisk/faxin/${FAXFILE},dfs) exten => h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten => h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)}) exten => h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten => h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten => h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten => h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten => h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)}) exten => h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)}) exten => h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)}) exten => h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)}) exten => h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)}) exten => h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)}) exten => h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)}) udptl.conf [general] udptlstart=4000 udptlend=4999 udptlfecentries = 3 udptlfecspan = 3 use_even_ports = no rtp.conf [general] rtpstart=1 rtpend=2 res_fax.conf [general] maxrate=14400 minrate=2400 statusevents=yes modems=v17,v27,v29 ecm=yes mail*CLI> core set verbose 6 Set remote console verbosity to 6 == Using SIP RTP CoS mark 5 -- Executing [41325122774@from-sip:1] Answer("SIP/sipcall.ch-008d", "") in new stack > 0x7f3964080f30 -- Probation passed - setting RTP source address to 123.456.789.123:20600 Got RTP packet from123.456.789.123:20600 (type 00, seq 042281, ts 1387619622, len 000160) -- Executing [41325122774@from-sip:2] Set("SIP/sipcall.ch-008d", "DB(lastcaller/number)=987654321") in new stack -- Executing [41325122774@from-sip:3] GotoIf("SIP/sipcall.ch-008d", "0?black,1") in new stack -- Executing [41325122774@from-sip:4] Wait("SIP/sipcall.ch-008d", "2") in new stack Got RTP packet from123.456.789.123:20600 (type 00, seq 042282, ts 1387619782, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042283, ts 1387619942, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042284, ts 1387620102, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042285, ts 1387620262, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042286, ts 1387620422, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042287, ts 1387620582, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042288, ts 1387620742, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042289, ts 1387620902, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042290, ts 1387621062, len 000160) Got RTP packet from123.456.789.123:20600 (type 00, seq 042291, ts 1387621222, len 000160