Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)
Le 18/02/2015 18:52, Eric Wieling a écrit : I solved the issue by not answering the call as I assume others have done. The only solution we found is to set faxdetect=no in the sip.conf of the peer definition. The Set(FAXOPT(detect)=[yes|no]) command in the dialplan is not taken in account. The problem with this solution is that we can't mix fax reception with direct fax line and fax detection on audio line. Or better said, to use a mix of those detection, we should put a Wait(x) in the fax direct DID and systematically use the fax extension to dial hylafax like [FAXDirectDID] exten = fax,1,Dial(IAX2/300,,) same = n,Hangup exten = _X.,1,Wait(10) same = n,Congestion() Not particulary clean. Daniel -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le 17/02/2015 17:00, Administrator TOOTAI a écrit : Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789@from-internal:1] Set(SIP/TOOTAi-8262, FAXOPT(faxdetect)=no) in new stack -- Executing [0123456789@from-internal:2] Macro(SIP/TOOTAi-8262, Fax) in new stack -- Executing [s@macro-Fax:1] Dial(SIP/TOOTAi-8262, IAX2/300,,) in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h@from-internal:1] Hangup(SIP/TOOTAi-8262, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/TOOTAi-8262' -- Hungup 'IAX2/300-7211' Thanks for your support No one have an idea on this ? -- _ -- 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] Res_fax - FAXOPT(faxdetect)
Le 18/02/2015 18:52, Eric Wieling a écrit : I solved the issue by not answering the call as I assume others have done. That's my problem: call is NOT answered :-( or better said, is answered by hylafax. That's why I thought that setting faxopt(faxdetect)=no would put asterisk out of the path. Asterisk version is 11.15.0 from Elastix. Same happend on a stock 11.16.0 Thanks for your answer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le 17/02/2015 17:00, Administrator TOOTAI a écrit : Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789@from-internal:1] Set(SIP/TOOTAi-8262, FAXOPT(faxdetect)=no) in new stack -- Executing [0123456789@from-internal:2] Macro(SIP/TOOTAi-8262, Fax) in new stack -- Executing [s@macro-Fax:1] Dial(SIP/TOOTAi-8262, IAX2/300,,) in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h@from-internal:1] Hangup(SIP/TOOTAi-8262, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/TOOTAi-8262' -- Hungup 'IAX2/300-7211' Thanks for your support No one have an idea on this ? -- _ -- 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] Res_fax - FAXOPT(faxdetect)
Hello Le 17/02/2015 17:00, Administrator TOOTAI a écrit : Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789@from-internal:1] Set(SIP/TOOTAi-8262, FAXOPT(faxdetect)=no) in new stack -- Executing [0123456789@from-internal:2] Macro(SIP/TOOTAi-8262, Fax) in new stack -- Executing [s@macro-Fax:1] Dial(SIP/TOOTAi-8262, IAX2/300,,) in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h@from-internal:1] Hangup(SIP/TOOTAi-8262, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/TOOTAi-8262' -- Hungup 'IAX2/300-7211' Thanks for your support No one have an idea on this ? -- Daniel -- _ -- 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] Res_fax - FAXOPT(faxdetect)
I solved the issue by not answering the call as I assume others have done. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le 17/02/2015 17:00, Administrator TOOTAI a écrit : Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789@from-internal:1] Set(SIP/TOOTAi-8262, FAXOPT(faxdetect)=no) in new stack -- Executing [0123456789@from-internal:2] Macro(SIP/TOOTAi-8262, Fax) in new stack -- Executing [s@macro-Fax:1] Dial(SIP/TOOTAi-8262, IAX2/300,,) in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h@from-internal:1] Hangup(SIP/TOOTAi-8262, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/TOOTAi-8262' -- Hungup 'IAX2/300-7211' Thanks for your support No one have an idea on this ? -- Daniel -- _ -- 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 -- _ -- 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] Res_fax - FAXOPT(faxdetect)
Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789@from-internal:1] Set(SIP/TOOTAi-8262, FAXOPT(faxdetect)=no) in new stack -- Executing [0123456789@from-internal:2] Macro(SIP/TOOTAi-8262, Fax) in new stack -- Executing [s@macro-Fax:1] Dial(SIP/TOOTAi-8262, IAX2/300,,) in new stack -- Called IAX2/300 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h@from-internal:1] Hangup(SIP/TOOTAi-8262, ) in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/TOOTAi-8262' -- Hungup 'IAX2/300-7211' Thanks for your support -- Daniel -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 24/10/2014 12:49 AM, Tim Nelson wrote: - Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. --Tim It would seem for Asterisk 11 and T.38 Gateway work for an IAX channel you require the following; IAX2 - SIP - T.38 Gateway - ITSP (SIP) Where as it would be nicer if it would accept acting as a gateway for an IAX channel i.e.; IAX2 - T.38 Gateway - ITSP (SIP) If an IAX2 channel is connected directly to a context with FAXOPT(t38gateway) enabled I see 'ast_rtp_read: RTP Read too short' messages and a failed transmission, the same is observed if using SIP with udptl=no instead of IAX2 channel; SIP (udptl=no) - T.38 Gateway - ITSP (SIP). Not sure if this is by design! Maybe time for another friendly chat with your ITSP in the hope they can resolve the issue. 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 25/10/2014 11:43 PM, Larry Moore wrote: On 24/10/2014 12:47 AM, Tim Nelson wrote: - Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim IAXmodem (other host on network) - Asterisk 1.2 (IAX) - Asterisk 1.8 with Fax Gateway Patch - SIP provider - PSTN Fax destination I have successfully sent a fax using a full page image via an Asterisk 1.2 system which forwards the request to my Asterisk 1.8 over an IAX channel, Asterisk 1.8 has the T.38 Fax Gateway patch installed. The outbound call triggered the T.38 gateway and the fax was received without error. I have ECM disabled in my IAX modem configuration in Hylafax. I don't have Asterisk 11 running to test with at this time however I confirmed the T.38 Gateway functions in Asterisk 11 when testing it. -- Accepting AUTHENTICATED call from 192.168.54.18: requested format = ulaw, requested prefs = (ulaw|alaw|slin), actual format = alaw, host prefs = (alaw|ulaw), priority = mine -- Executing [PSTN Number@FAX-T30:1] Dial(IAX2/faxgw-iax-1210, SIP/PSTN Number@itsp-fax,55) in new stack == Using SIP RTP TOS bits 184 -- Called SIP/PSTN Number@itsp-fax -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 == Using SIP RTP TOS bits 184 -- SIP/itsp-fax-000b answered IAX2/faxgw-iax-1210 [Oct 25 23:24:11] NOTICE[27896]: channel.c:4220 __ast_read: Dropping incompatible voice frame on IAX2/faxgw-iax-1210 of format slin since our native format has changed to 0x8 (alaw) -- Got Fax Tone CED Chan SIP/itsp-fax-000b [1] Sending T.38 Params Peer Is IAX2/faxgw-iax-1210 [0] -- Request on IAX2/faxgw-iax-1210 [0] Storing I: SIP/itsp-fax-000b [1] == Using UDPTL TOS bits 184 -- Negotiated on SIP/itsp-fax-000b [4] Ignoring I: IAX2/faxgw-iax-1210 [0] -- T.38 Gateway starting for chan SIP/itsp-fax-000b and peer IAX2/faxgw-iax-1210 pbx*CLI iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format FirstMsg LastMsg IAX2/faxgw-iax-1210 192.168.54.18 faxgw-iax 01210/4 00010/5 0ms -0001ms ms alaw Rx:NEW Tx:ACK 1 active IAX channel pbx*CLI fax show sessions Current FAX Sessions: Channel Tech FAXID Type Operation State File(s) SIP/itsp-fax-000 Spandsp 1 T.38 receive Active (null) 1 FAX sessions -- Executing [h@FAX-T30:1] GotoIf(IAX2/faxgw-iax-1210, 0?2:3) in new stack -- Goto (FAX-T30,h,3) -- Executing [h@FAX-T30:3] NoOp(IAX2/faxgw-iax-1210, Finish if_FAX-T30_37) in new stack -- Executing [h@FAX-T30:4] NoOp(IAX2/faxgw-iax-1210, Call/Fax Ended 2014-10-25 23:27:38 +0800) in new stack -- Connection Statistics Bit Rate :14400 ECM : No Pages : 1 == Spawn extension (FAX-T30, PSTN Number, 1) exited non-zero on 'IAX2/faxgw-iax-1210' -- Hungup 'IAX2/faxgw-iax-1210' Well, forgive me as I should have had an Asterisk 11 system up and running and performing tests before posting. It would appear there is a behavioural difference with the patch created for Asterisk 1.8 and the implementation applied to Asterisk 11. The observations as listed above relating to the fax gateway stepping in, occurs when an outbound fax call is made using either of the g711 codecs, Asterisk detects the fax tones in the calling leg about 3 seconds after the call has been answered and sends a T.38 re-invite to the ITSP. Using Asterisk 11, when an outbound call is made, the fax gateway detection feature does not do anything on the leg of the call (as you have observed) to the ITSP until it receives a T.38 re-invite from the ITSP, my observations show this occurs about 4 seconds after the call is answered. I suspect once the T.38 re-invite is received from the ITSP, the T.38 Gateway sends a T.38 re-invite on the leg of the caller to check if it is capable of T.38. I have not confirmed this definitively. I'm obviously fortunate my ITSP is correctly handling 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 24/10/2014 12:47 AM, Tim Nelson wrote: - Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim IAXmodem (other host on network) - Asterisk 1.2 (IAX) - Asterisk 1.8 with Fax Gateway Patch - SIP provider - PSTN Fax destination I have successfully sent a fax using a full page image via an Asterisk 1.2 system which forwards the request to my Asterisk 1.8 over an IAX channel, Asterisk 1.8 has the T.38 Fax Gateway patch installed. The outbound call triggered the T.38 gateway and the fax was received without error. I have ECM disabled in my IAX modem configuration in Hylafax. I don't have Asterisk 11 running to test with at this time however I confirmed the T.38 Gateway functions in Asterisk 11 when testing it. -- Accepting AUTHENTICATED call from 192.168.54.18: requested format = ulaw, requested prefs = (ulaw|alaw|slin), actual format = alaw, host prefs = (alaw|ulaw), priority = mine -- Executing [PSTN Number@FAX-T30:1] Dial(IAX2/faxgw-iax-1210, SIP/PSTN Number@itsp-fax,55) in new stack == Using SIP RTP TOS bits 184 -- Called SIP/PSTN Number@itsp-fax -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 -- SIP/itsp-fax-000b is making progress passing it to IAX2/faxgw-iax-1210 == Using SIP RTP TOS bits 184 -- SIP/itsp-fax-000b answered IAX2/faxgw-iax-1210 [Oct 25 23:24:11] NOTICE[27896]: channel.c:4220 __ast_read: Dropping incompatible voice frame on IAX2/faxgw-iax-1210 of format slin since our native format has changed to 0x8 (alaw) -- Got Fax Tone CED Chan SIP/itsp-fax-000b [1] Sending T.38 Params Peer Is IAX2/faxgw-iax-1210 [0] -- Request on IAX2/faxgw-iax-1210 [0] Storing I: SIP/itsp-fax-000b [1] == Using UDPTL TOS bits 184 -- Negotiated on SIP/itsp-fax-000b [4] Ignoring I: IAX2/faxgw-iax-1210 [0] -- T.38 Gateway starting for chan SIP/itsp-fax-000b and peer IAX2/faxgw-iax-1210 pbx*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format FirstMsgLastMsg IAX2/faxgw-iax-1210 192.168.54.18faxgw-iax 01210/4 00010/5 0ms -0001ms ms alawRx:NEW Tx:ACK 1 active IAX channel pbx*CLI fax show sessions Current FAX Sessions: Channel Tech FAXID Type Operation State File(s) SIP/itsp-fax-000 Spandsp1 T.38 receiveActive (null) 1 FAX sessions -- Executing [h@FAX-T30:1] GotoIf(IAX2/faxgw-iax-1210, 0?2:3) in new stack -- Goto (FAX-T30,h,3) -- Executing [h@FAX-T30:3] NoOp(IAX2/faxgw-iax-1210, Finish if_FAX-T30_37) in new stack -- Executing [h@FAX-T30:4] NoOp(IAX2/faxgw-iax-1210, Call/Fax Ended 2014-10-25 23:27:38 +0800) in new stack -- Connection Statistics Bit Rate :14400 ECM : No Pages : 1 == Spawn extension (FAX-T30, PSTN Number, 1) exited non-zero on 'IAX2/faxgw-iax-1210' -- Hungup 'IAX2/faxgw-iax-1210' -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ -Dave -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 23/10/2014 3:55 AM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now. -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ It is up now, thanks! --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 3:55 AM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now. The version of SpanDSP is not in question at this point. The problem lies in I need a way to use the T38 Gateway function, but *also* initiate the reinvite to T.38 on the call as the provider will not do this, saying it is the *caller*'s responsibility. This is contrary to past experience however... --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 24/10/2014 12:49 AM, Tim Nelson wrote: - Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. The canreinvite= option is an old setting, this is replaced by the directmedia= option in newer versions of Asterisk, it doesn't prevent a re-invite, it keeps the audio going through asterisk rather than negotiating an audio channel directly with the other endpoint. The reason I suggested disabling udptl at that end is because my understanding of how the implementation of T.38 Gateway works on Asterisk is; 1) it does not utilise any of the T.38 gateway features in spandsp 2) the gateway will not step in if the originator negotiates T.38 Considering the other post you sent, are you suing IAX between the two Asterisk boxes? To test the T.38 Gateway can work on your box in question set up an IAX modem and configure HylaFAX modem to use the iaxmodem on the box in question, test the gateway functionality. When I tested Asterisk 11 a little while back I configured HylaFAX on my current system to communicate with an IAX modem on my Asterisk 11 test box and was able to observe the T.38 gateway function. I can't tell from the information you've provided if the old Asterisk box is on the same network or having to traverse a WAN link to make the connection out through to your SIP provider. Perhaps you could provide more information about your set up such as entries from your sip.conf, iax.conf, udptl.conf etc. 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim -- _ -- 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] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
Greetings- Working with the T.38 gateway functionality that is sparsely documented [1] , I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2] . Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 3:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). _ Kevin The 304599 rev does seem to work good. I just finished my testing on it and the F option works great. I have three more test to do and if they pass it should be good to go. When could it get into the releases? Thanks Bryant -- _ -- 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] res_fax
On 01/31/2011 02:08 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Thursday, January 27, 2011 3:08 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). _ Kevin The 304599 rev does seem to work good. I just finished my testing on it and the F option works great. I have three more test to do and if they pass it should be good to go. When could it get into the releases? It's a new feature, so it won't go into any existing release branches; the first release that will have this addition is Asterisk 1.10.1. Of course, the patch is quite small as you've seen, so it will be easy for you to apply it to your installations. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Monday, January 31, 2011 5:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/31/2011 02:08 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Thursday, January 27, 2011 3:08 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). _ Kevin The 304599 rev does seem to work good. I just finished my testing on it and the F option works great. I have three more test to do and if they pass it should be good to go. When could it get into the releases? It's a new feature, so it won't go into any existing release branches; the first release that will have this addition is Asterisk 1.10.1. Of course, the patch is quite small as you've seen, so it will be easy for you to apply it to your installations. _ Kevin I just replaced the res_fax.c file with the one from 304599. Would I just keep doing that as I step forward on versions of 1.8.x? If this is the case how would I get any other critical changes to res_fax.c that may occur after rev 304599? How would I create a patch that would allow me to apply it to additional release version of asterisk. Sorry for the simple questions I do most of my dev on windows machines and this process is a still a bit confusing to me. Thanks Bryant -- _ -- 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] res_fax
On 01/31/2011 05:06 PM, Bryant Zimmerman wrote: I just replaced the res_fax.c file with the one from 304599. Would I just keep doing that as I step forward on versions of 1.8.x? If this is the case how would I get any other critical changes to res_fax.c that may occur after rev 304599? How would I create a patch that would allow me to apply it to additional release version of asterisk. Sorry for the simple questions I do most of my dev on windows machines and this process is a still a bit confusing to me. It's very possible that future versions of res_fax.c from trunk will not be compatible with Asterisk 1.8.x, so you can't keep doing that forever. However, as long as the version of res_fax.c *compiles* when you drop it into the Asterisk 1.8 tree, it should work. Critical changes to res_fax.c (meaning bug fixes or security vulnerability fixes) *will* be made in the 1.8 branch; it's only new features that won't be added there. I can't really tell you how you might want to make patches and apply them to future releases... that depends a lot on how you are downloading and building Asterisk 1.8 already. Since the patch to add 'F' is so small, though, it would be fairly easy to manually make the changes when you install a new release. Now that you have a working system, it would be really nice to get some debugging information like I asked for before, so we can try to figure out why T.38 negotiation is failing with your provider. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Thanks Bryant -- _ -- 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] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? My schedule is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 10:31 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? My schedule is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin Thanks I am continuing with other parts of my fax code as well for now. I will test any changes as you are able to make them. Bryant -- _ -- 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] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Thursday, January 27, 2011 3:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/27/2011 09:21 AM, Bryant Zimmerman wrote: Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 Kevin I tried everthing I could think of to get the n option to work last night but it would not do a complete shut off of the T.38 option and would not receive a fax. What do you need from me on the debug side so I can help you get it working as expected? Revision 304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). - Kevin I will rebuild and test in a bit. Thanks Bryant -- _ -- 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] res_fax
Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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] res_fax
On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Steve did not write res_fax (which where SendFAX and ReceiveFAX come from) snip I am personally a little confused here, because I have a ReceiveFAX application when I unload the res_fax module and res_fax_digium module and load the app_fax module. In other words, I think that multiple modules provide applications named ReceiveFax and SendFAX. Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? Tom -- _ -- 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] res_fax
On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Steve did not write res_fax (which where SendFAX and ReceiveFAX come from) snip I am personally a little confused here, because I have a ReceiveFAX application when I unload the res_fax module and res_fax_digium module and load the app_fax module. In other words, I think that multiple modules provide applications named ReceiveFax and SendFAX. Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 1:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt cap-t38.pcap And by the way thank you for your response it is appreciated. Thanks Bryant Zimmerman (ZK Tech Inc.) -- _ -- 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] res_fax
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote: On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. Gotcha. So, 1.6 users who install FFA get res_fax and res_fax_digium. Presumably, 1.6 users could also combine res_fax and res_fax_spandsp? Steve - Will compiling the latest version of SpanDSP on a 1.6 system result in a res_fax_spandsp.so module? Tom -- _ -- 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] res_fax
On 01/26/2011 01:21 PM, Tom Rymes wrote: On 01/26/2011 2:16 PM, Kevin P. Fleming wrote: On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now the way to go? That is correct. app_fax is deprecated (and that is why it is marked as don't build by default), and res_fax plus a technology module (res_fax_spandsp or res_fax_digium) is the replacement for it. All of the work that the Digium team has done improving T.38 negotiation and interoperability has gone into res_fax, not app_fax. Users of Asterisk 1.8.x should only choose to build app_fax if they have a specific need for it (and if that's the case we'd like to know what the need is so we can ensure that res_fax can satisfy it). Users of older Asterisk releases will have app_fax by default (since res_fax was not included in those versions), but if they want to use Digium's res_fax_digium module they can download it along with res_fax and use them instead. Gotcha. So, 1.6 users who install FFA get res_fax and res_fax_digium. Presumably, 1.6 users could also combine res_fax and res_fax_spandsp? Steve - Will compiling the latest version of SpanDSP on a 1.6 system result in a res_fax_spandsp.so module? No, res_fax_spandsp is not part of SpanDSP (but it uses SpanDSP), and we don't distribute an Asterisk 1.6.x version of res_fax_spandsp.c. It wouldn't be hard for someone to make one, though, and the res_fax binary module download does include res_fax.h so it is possible to compile against it if they wanted to do so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt http://webmail.zktech.com/public/downloadfile.aspx?f=KERoF6PWf6e2FK8S5zgEDs02rFGdd7zE0AIG7tjbCR9a06oFY1NwFap58zDWva3BcdOp%2b%2f%2fuBo8%3d cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljmT%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d Unfortunately that log capture is incomplete; it doesn't include any of the messages that res_fax emits as it goes through T.38 negotiations. Please ensure that your 'console' channel in logger.conf has 'debug,verbose,warning,notice,error,fax' enabled and that you have 'core set verbose 10' and 'core set debug 10' set before the call attempt begins (or at least before ReceiveFAX is executed). If the server is only processing this particular call, then 'sip set debug on' would also be helpful. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 2:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 12:42 PM, Bryant Zimmerman wrote: Steve Are there any undocumented options available with ReceiveFAX and the res_fax_spandsp module. I am having issues with getting t.38 to negotiate with Level 3 faxes but if I force t.30 the fax comes in. But the fax does not fall back t.30 if the t.38 fails You haven't posted any logs of the failing attempts, or packet captures of the SIP traffic, so it's pretty much impossible for anyone to help you debug this (anyone who tried would just be guessing). Steve did not write res_fax (which where SendFAX and ReceiveFAX come from), and there are no 'undocumented' options available for it, because it's open source and the source code shows all the options that are available. If you would like to try to figure out what is going on, start by posting a *complete* log file from Asterisk for a failed inbound FAX attempt, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin These were attached to another post. Here are the links again Fax Debug.txt http://webmail.zktech.com/public/downloadfile.aspx?f=KERoF6PWf6e2FK8S5zgEDs 02rFGdd7zE0AIG7tjbCR9a06oFY1NwFap58zDWva3BcdOp%2b%2f%2fuBo8%3d cap-t38.pcap http://webmail.zktech.com/public/downloadfile.aspx?f=ulHIhepag5qoKm0cTUmljm T%2f7YCcOPvzlyZcnZg%2fG2B25W%2fsSr6Uwbu%2bET3kbKw84pTJjtuqrPQ%3d Unfortunately that log capture is incomplete; it doesn't include any of the messages that res_fax emits as it goes through T.38 negotiations. Please ensure that your 'console' channel in logger.conf has 'debug,verbose,warning,notice,error,fax' enabled and that you have 'core set verbose 10' and 'core set debug 10' set before the call attempt begins (or at least before ReceiveFAX is executed). If the server is only processing this particular call, then 'sip set debug on' would also be helpful. - Kevin I will get the additional debugs done when there is no other load on the fax. Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 Thanks Bryant -- _ -- 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] res_fax
On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 4:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Thanks Bryant -- _ -- 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] res_fax
From: Kevin P. Fleming kpflem...@digium.com Sent: Wednesday, January 26, 2011 5:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_fax On 01/26/2011 04:16 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 4:52 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Most definitely; I can see cases like yours where someone would want to be able to forcibly disable T.38 for specific calls for troubleshooting purposes. In fact... if you give me about 15 minutes, I'll commit a patch to Asterisk trunk to add an option to do that, and you can backport it to the version you are using :-) Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? Much thanks on this. Bryant -- _ -- 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] res_fax
On 01/26/2011 04:36 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 5:21 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 04:16 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 4:52 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01/26/2011 03:14 PM, Bryant Zimmerman wrote: Is there a way for me to force t.38 off for a call but to allow t.38 for other calls. What I am thinking is if a t.38 fails to flag the next call from that number to g711 audio. This would at least let me work arround the issue for now where t.38 fails with some endpoints but not others and the g711 audio will work. The issue I am seeing is it appears that with some endpoinds on Level 3 that the t.38 tunnel comes up fine but no fax data starts flowing but this only is happening with faxes coming from some Cisco gateways sending out via PRI using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. With out this I have no way to force the fall back then and the faxes will always fail in this case because t38 successfully negotiates.. Do you have any other ideas? If I pick arround in the source what might it take to add another option to the ReceiveFAX to only do g711 audio? Is this somthing that I could get submitted back into the tree if I can figure it out? Most definitely; I can see cases like yours where someone would want to be able to forcibly disable T.38 for specific calls for troubleshooting purposes. In fact... if you give me about 15 minutes, I'll commit a patch to Asterisk trunk to add an option to do that, and you can backport it to the version you are using :-) Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
Kevin That is grate. I dove into the code and tried to add it my self I added a F option but I have not figured out the right spot to force the exclusion to shut off the T38. Where will the patch be posted? http://svnview.digium.com/svn/asterisk?view=revrev=304342 - Kevin I downloaded 1.8.2.3 and copied the modified version of res_fax.c into my the res folder. I built and installed the version of asterisk. When I use the new n option with ReceiveFAX I get a bunch of WARNING messages on the console that state. [Jan 26 20:43:38] WARNING[23393]: chan_sip.c:6047 sip_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) If I shut of the n option it goes back to the normal behavior. It appears that there is somthing missing in the n option and it is not causing it to fall back to audio only mode. as it would if t38pt_udptl=no Bryant -- _ -- 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] res_fax
On 01/21/2011 08:37 PM, Tom Rymes wrote: On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote: A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. As a longtime Hylafax user, I can confirm it's an excellent solution. I am somewhat surprised about the comment of being able to do audio or t.38, but not both. This is probably a little true and untrue at the same time, though I have never used t.38modem with Hylafax. Given the structure of the product, you could have HylaFAX connected to both an IAXModem and a T.38Modem at the same time (or 23 IAXModems, a 24-port T1/E1 PCI-card modem, and 7 t.38modems for that matter...). What it cannot do, is receive audio and t.38 on the same port, which is what I presume that Steve was referring to. This is really a limitation of IAXmodem and t.38modem, as one only handles audio, the other only handles t.38. In other words, you could route t.38 faxes to it on port 1 and audio faxes on port2, but you cannot have port 1 handle both types. Its easy to set up some t38modem channels and some iaxmodem channels for receiving FAXes. Transmit is more problematic. With this split config, you need to know in advance whether the particular number is accessible by T.38 or by audio. Most people won't. Steve -- _ -- 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] res_fax
On 01/21/2011 8:59 AM, Steve Underwood wrote: On 01/21/2011 08:37 PM, Tom Rymes wrote: On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote: [snip] Its easy to set up some t38modem channels and some iaxmodem channels for receiving FAXes. Transmit is more problematic. With this split config, you need to know in advance whether the particular number is accessible by T.38 or by audio. Most people won't. Steve Good point. Perhaps you could route via chan_clairvoyant? Tom -- _ -- 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] res_fax
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] res_fax
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve -- _ -- 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] res_fax
On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? -- _ -- 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] res_fax
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve -- _ -- 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] res_fax
On Jan 20, 2011, at 8:53 PM, Steve Underwood On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve Steve thanks for your response. Do I need a copy of spandsp installed or is the res_fax_spandsp.so the complete package. If I need spandsp what version should I be using? The version I compiled and am using is now over a year old spandsp-0.0.5pre4. Where can I get the current stable version with a list of dependencies for compilation? Thanks Bryant -- _ -- 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] res_fax
On Jan 19, 2011, at 3:18 PM, Jason Parker wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Am I the only one confused here? (probably) It seems like you imply that SendeFax (which looks like a typo to me) is correct in the second sentence, then reverse yourself in the last parenthetical statement. Tom -- _ -- 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] res_fax
There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Am I the only one confused here? (probably) It seems like you imply that SendeFax (which looks like a typo to me) is correct in the second sentence, then reverse yourself in the last parenthetical statement. I'm not confused if he means that the content of Application_SendeFax is correct and the content of Application_SendFax is old. After the next update, the content of Application_SendFax will be correct and Application_SendeFax will go away --Don -- _ -- 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
