fax по t38

/etc/asterisk/udptl.conf

;
; UDPTL Configuration (UDPTL is one of the transports for T.38)
;
[general]
;
; UDPTL start and UDPTL end configure start and end addresses
;
udptlstart=4000
udptlend=4999
;
; Whether to enable or disable UDP checksums on UDPTL traffic
;
;udptlchecksums=no
;
; The number of error correction entries in a UDPTL packet
;
udptlfecentries = 3
;
; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; Some VoIP providers will only accept an offer with an even-numbered
; UDPTL port. Set this option so that Asterisk will only attempt to use
; even-numbered ports when negotiating T.38. Default is no.
use_even_ports = no

<------------>
    -- Executing [74956986112@from-trunk-sip-outComtelko:1] Set("SIP/outComtelko-00000001", "GROUP()=OUT_1") in new stack
    -- Executing [74956986112@from-trunk-sip-outComtelko:2] Goto("SIP/outComtelko-00000001", "from-trunk,74956986112,1") in new stack
    -- Goto (from-trunk,74956986112,1)
    -- Executing [74956986112@from-trunk:1] Set("SIP/outComtelko-00000001", "__FROM_DID=74956986112") in new stack
    -- Executing [74956986112@from-trunk:2] ExecIf("SIP/outComtelko-00000001", "0 ?Set(CALLERID(name)=+12314987446)") in new stack
    -- Executing [74956986112@from-trunk:3] Set("SIP/outComtelko-00000001", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [74956986112@from-trunk:4] Set("SIP/outComtelko-00000001", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [74956986112@from-trunk:5] Goto("SIP/outComtelko-00000001", "ext-featurecodes,666,1") in new stack
    -- Goto (ext-featurecodes,666,1)
    -- Executing [666@ext-featurecodes:1] Goto("SIP/outComtelko-00000001", "from-internal,666,1") in new stack
    -- Goto (from-internal,666,1)
    -- Executing [666@from-internal:1] Set("SIP/outComtelko-00000001", "FAX_RX_EMAIL=artec@agava.com") in new stack
    -- Executing [666@from-internal:2] Goto("SIP/outComtelko-00000001", "ext-fax,s,1") in new stack
    -- Goto (ext-fax,s,1)
    -- Executing [s@ext-fax:1] Macro("SIP/outComtelko-00000001", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/outComtelko-00000001", "AMPUSER=+12314987446") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/outComtelko-00000001", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/outComtelko-00000001", "1?Set(REALCALLERIDNUM=+12314987446)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/outComtelko-00000001", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/outComtelko-00000001", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/outComtelko-00000001", "1?report") in new stack
    -- Goto (macro-user-callerid,s,11)
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/outComtelko-00000001", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/outComtelko-00000001", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/outComtelko-00000001", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,24)
    -- Executing [s@macro-user-callerid:24] Set("SIP/outComtelko-00000001", "CALLERID(number)=+12314987446") in new stack
    -- Executing [s@macro-user-callerid:25] Set("SIP/outComtelko-00000001", "CALLERID(name)=12314987446") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/outComtelko-00000001", "CHANNEL(language)=en") in new stack
    -- Executing [s@ext-fax:2] NoOp("SIP/outComtelko-00000001", "Receiving Fax for: artec@agava.com , From: "12314987446" <+12314987446>") in new stack
    -- Executing [s@ext-fax:3] StopPlayTones("SIP/outComtelko-00000001", "") in new stack
    -- Executing [s@ext-fax:4] ReceiveFAX("SIP/outComtelko-00000001", "/var/spool/asterisk/fax/1324457361.1.tif,f") in new stack
    -- Channel 'SIP/outComtelko-00000001' receiving FAX '/var/spool/asterisk/fax/1324457361.1.tif'
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.231.214.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.231.214.10:5060;branch=z9hG4bK-10d8a328-4ef19d90;vtservice=CallControl.CallControlServlet;received=91.231.214.10;rport=5060
From: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=26188
To: <sip:74956986112@89.108.96.17:53039>;tag=as6c525f9d
Call-ID: 477CB1F7-5C9E-426D-82A9-514C0E462A23
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74956986112@192.168.0.4:5060>
Content-Type: application/sdp
Content-Length: 282

Смущают эти строки Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP т.е. факс пытается всетаки через g711 пройти, а у них так не ходит, как сделать, чтобы через t38 ходило?

не пойму как описать? Я поднял астериск 192.168.0.4 с транком от провайдера comtelco номер 6986112, настроил там факсы, посылаю на этот номер факс, слышу писк факса, нажимаю старт. И после этого вылезает такой дебаг.

Благодаря создателю статьи

Проблема решилась в sip_general_custom.conf необходимо добавить следующее :

t38pt_udptl=yes,redundancy,maxdatagram=400
faxdetect=yes
echocancel=yes

Единственное теперь надо что то придумать с кач-вом изображения, де видел что можно меня его.C:\fakepath\1324544938.4.jpg