hutx
2009-Jul-16 13:51 UTC
[asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1
I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax
(.tiff) from the first
asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an
INVITE with audio G.711. Asterisk2
accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to
Asterisk1. But, Asterisk1 responds with
"488 not acceptable here". I double check t38pt_udptl = yes in my
sip.conf. Why not Asterisk1 can not accept the Re-INVITE
with T.38 SDP? What do I miss?
////////////////////////////////////////////////////////////////////
dev10*CLI> fax show version
Fax For Asterisk Components:
dev10*CLApplications: 1.6.1_1.0.11
dev10*CLDigium Fax T.38 Driver: 1.6.1_1.0.9 (optimized for i686_32)
dev10*CLDigium Fax G.711 Driver: 1.6.1_1.0.9 (optimized for i686_32)
----------------------------------------------------------
.call file
Channel: SIP/1905 at outbound-calls
MaxRetries: 3
WaitTime: 30
Set: LOCALSTATIONID=22222
Set: LOCALHEADERINFO=T38 fax
Set: T38CALL=1
Set: T38TXDETECT=yes
CallerID: 123456
Context: fax-tx
Extension: send
priority:1
-----------------------------------------------------------
sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
t38pt_udptl = yes
------------------------------------------------------------
extensions.conf
[fax-rx]
exten => receive,1,NoOp(**** FAX RECEIVE ****)
exten => receive,n,Set(GLOBAL(FAXCOUNT)=${GLOBAL(FAXCOUNT)}+1)
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif)
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten => receive,n,NoOp(**** SETTING FAXOPT ****)
exten => receive,n,Set(FAXOPT(ecm)=yes)
exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
exten => receive,n,Set(FAXOPT(localstationid)=1234567890)
exten => receive,n,Set(FAXOPT(maxrate)=14400)
exten => receive,n,Set(FAXOPT(minrate)=2400)
exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten => receive,n,NoOp(**** RECEIVING FAX : ${FAXFILE} ****)
exten => receive,n,ReceiveFAX(/home/sip/fax/${FAXFILE})
[fax-tx]
exten => send,1,NoOp(**** SENDING FAX ****)
exten => send,n,Wait(6)
exten => send,n,Set(GLOBAL(FAXCOUNT)=1)
;exten => send,n,Set(GLOBAL(FAXCOUNT)= ${GLOBAL(FAXCOUNT)}+1)
exten => send.,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => send,n,Set(FAXFILE=test.tif)
; Set FAXOPTs
exten => send,n,NoOp(**** SETTING FAXOPT ****)
exten => send,n,Set(FAXOPT(ecm)=yes)
exten => send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)}
at ${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten => send,n,Set(FAXOPT(localstationid)=1234567890)
exten => send,n,Set(FAXOPT(maxrate)=14400)
exten => send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten => send,n,NoOp(**** SENDING FAX : ${FAXFILE} ****)
exten => send,n,SendFAX(/home/sip/fax/${FAXFILE},d)
[default]
exten => _X.,1,NoOp(**** FAX DETECTED ****)
exten => _X.,n,Goto(fax-rx,receive,1)
----------------------------------------------------------
The SIP trace is
#
U 2009/07/15 22:30:11.588436 74.13.233.143:5060 -> 209.167.0.151:5060
INVITE sip:1905 at 209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK092e48ce;rport..Max-Forwards: 70..From
: "123456" <sip:123456 at 74.13.233.143>;tag=as74992a24..To:
<sip:1905 at 209.167.0.151>..Contact: <sip:123456 at
74.13.233.143>..Call-I
D: 422fd4375fe79a5977e891870f5cc05b at 74.13.233.143..CSeq: 102
INVITE..User-Agent: Asterisk PBX 1.6.1.1..Date: Wed, 15 Jul 2009 2
2:30:11 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY..Supported: replaces, timer..Content-Type: appl
ication/sdp..Content-Length: 265....v=0..o=root 1425900082 1425900082 IN IP4
74.13.233.143..s=Asterisk PBX 1.6.1.1..c=IN IP4 74
.13.233.143..t=0 0..m=audio 18452 RTP/AVP 0 101..a=rtpmap:0
PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=si
lenceSupp:off - - - -..a=ptime:20..a=sendrecv..
#
U 2009/07/15 22:30:11.723006 209.167.0.151:5060 -> 74.13.233.143:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From:
"123456
" <sip:123456 at 74.13.233.143>;tag=as74992a24..To: <sip:1905 at
209.167.0.151>..Call-ID: 422fd4375fe79a5977e891870f5cc05b at 74.13.233.
143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Suppor
ted: replaces, timer..Contact: <sip:1905 at
209.167.0.151>..Content-Length: 0....
#
U 2009/07/15 22:30:11.730205 209.167.0.151:5060 -> 74.13.233.143:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From:
"123456" <s
ip:123456 at 74.13.233.143>;tag=as74992a24..To: <sip:1905 at
209.167.0.151>;tag=as3114c7a3..Call-ID: 422fd4375fe79a5977e891870f5cc05b
@74.13.233.143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOT
IFY..Supported: replaces, timer..Contact: <sip:1905 at
209.167.0.151>..Content-Type: application/sdp..Content-Length: 265....v=0..
o=root 2128364626 2128364626 IN IP4 209.167.0.151..s=Asterisk PBX
1.6.1.1..c=IN IP4 209.167.0.151..t=0 0..m=audio 13848 RTP/AVP
0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - - -..a=ptime:20..a=send
recv..
#
U 2009/07/15 22:30:11.730460 74.13.233.143:5060 -> 209.167.0.151:5060
ACK sip:1905 at 209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP
74.13.233.143:5060;branch=z9hG4bK271a4788;rport..Max-Forwards: 70..From: "
123456" <sip:123456 at 74.13.233.143>;tag=as74992a24..To:
<sip:1905 at 209.167.0.151>;tag=as3114c7a3..Contact: <sip:123456 at
74.13.233.
143>..Call-ID: 422fd4375fe79a5977e891870f5cc05b at 74.13.233.143..CSeq: 102
ACK..User-Agent: Asterisk PBX 1.6.1.1..Content-Length:
0....
#
U 2009/07/15 22:30:12.140990 209.167.0.151:5060 -> 74.13.233.143:5060
INVITE sip:123456 at 74.13.233.143 SIP/2.0..Via: SIP/2.0/UDP
209.167.0.151:5060;branch=z9hG4bK00000000;rport..Max-Forwards: 70..Fr
om: <sip:1905 at 209.167.0.151>;tag=as3114c7a3..To: "123456"
<sip:123456 at 74.13.233.143>;tag=as74992a24..Contact: <sip:1905 at
209.167
.0.151>..Call-ID: 422fd4375fe79a5977e891870f5cc05b at 74.13.233.143..CSeq:
102 INVITE..User-Agent: Asterisk PBX 1.6.1.1..Allow: IN
VITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
replaces, timer..Content-Type: application/sdp..Content-L
ength: 288....v=0..o=root 2128364626 2128364627 IN IP4
209.167.0.151..s=Asterisk PBX 1.6.1.1..c=IN IP4 209.167.0.151..t=0 0..m image
4240 udptl
t38..a=T38FaxVersion:0..a=T38MaxBitRate:9600..a=T38FaxRateManagement:transferredTCF..a=T38FaxMaxBuffer:400..a
T38FaxMaxDatagram:400..a=T38FaxUdpEC:t38UDPFEC..
#
U 2009/07/15 22:30:12.141353 74.13.233.143:5060 -> 209.167.0.151:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
209.167.0.151:5060;branch=z9hG4bK00000000;received=209.167.0.151;rport=5060..From:
<sip:19
05 at 209.167.0.151>;tag=as3114c7a3..To: "123456" <sip:123456
at 74.13.233.143>;tag=as74992a24..Call-ID: 422fd4375fe79a5977e891870f5c
c05b at 74.13.233.143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY..Supported: replaces, timer..Contact: <sip:123456 at
74.13.233.143>..Content-Length: 0....
#
U 2009/07/15 22:30:17.140683 74.13.233.143:5060 -> 209.167.0.151:5060
SIP/2.0 488 Not acceptable here..Via: SIP/2.0/UDP
209.167.0.151:5060;branch=z9hG4bK00000000;received=209.167.0.151;rport=5060..
From: <sip:1905 at 209.167.0.151>;tag=as3114c7a3..To: "123456"
<sip:123456 at 74.13.233.143>;tag=as74992a24..Call-ID: 422fd4375fe79a5
977e891870f5cc05b at 74.13.233.143..CSeq: 102 INVITE..Server: Asterisk PBX
1.6.1.1..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE
R, SUBSCRIBE, NOTIFY..Supported: replaces, timer..Content-Length:
0..X-Asterisk-HangupCause: Normal Clearing..X-Asterisk-Hangup
CauseCode: 16....
#
U 2009/07/15 22:30:17.186266 209.167.0.151:5060 -> 74.13.233.143:5060
ACK sip:123456 at 74.13.233.143 SIP/2.0..Via: SIP/2.0/UDP
209.167.0.151:5060;branch=z9hG4bK00000000;rport..Max-Forwards: 70..From:
<sip:1905 at 209.167.0.151>;tag=as3114c7a3..To: "123456"
<sip:123456 at 74.13.233.143>;tag=as74992a24..Contact: <sip:1905 at
209.167.0.
151>..Call-ID: 422fd4375fe79a5977e891870f5cc05b at 74.13.233.143..CSeq: 102
ACK..User-Agent: Asterisk PBX 1.6.1.1..Content-Length:
0....
#
------------------------------------------------------------
Kevin P. Fleming
2009-Jul-16 14:30 UTC
[asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1
hutx wrote:> I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first > asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 > accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with > "488 not acceptable here". I double check t38pt_udptl = yes in my sip.conf. Why not Asterisk1 can not accept the Re-INVITE > with T.38 SDP? What do I miss?You need to post a console log from the Asterisk console of this problem occurring, with 'core set verbose 10', 'core set debug 10' and 'sip set debug on' (and ensure that the 'debug' logger level is activated for the console log channel in logger.conf). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpfleming at digium.com Check us out at www.digium.com & www.asterisk.org