Asterisk 1.6.x and t.38

Since long time I didn’t checked current support of t38 in aster

So, recently I saw that it is suppose to be in 1.6 and in patched 1.4.x
Tried to go with 1.6.1 first and last spandsp-0.0.6

Compiled all stuff with no issues
And tried to send fax with Zopier

No luck.
Same with spandsp 0.0.5
Same issue with aster 1.6.0.9
Not played with 1.4 yet.

Sure I’ve enabled t38pt_udptl = yes in sip conf.
Only ulaw from both sides Zopier and asterisk.

Was searching all day but no solution.

Really stuck here.

Having following scenario.

[size=75]Asterisk 1.6.0.9, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 1.6.0.9 currently running on tank (pid = 29195)
Verbosity is at least 100
tankCLI>
tank
CLI>
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
INVITE sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:test@10.10.0.188:5060;transport=UDP
To: sip:100@10.10.0.157;transport=UDP
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Content-Length: 201

v=0
o=Zoiper_user 0 0 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (12 headers 10 lines) —
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Sending to 10.10.0.188 : 5060 (NAT)
Using INVITE request as basis request - ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Found user ‘test’ for ‘test’

<— Reliably Transmitting (no NAT) to 10.10.0.188:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP;tag=as38d974a0
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2b2a44fc"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.’ in 32000 ms (Method: INVITE)
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
ACK sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1—d8754z-;rport
To: sip:100@10.10.0.157;transport=UDP;tag=as38d974a0
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
INVITE sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:test@10.10.0.188:5060;transport=UDP
To: sip:100@10.10.0.157;transport=UDP
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username=“test”,realm=“asterisk”,nonce=“2b2a44fc”,uri="sip:100@10.10.0.157;transport=UDP",response=“977211f
ce0fadb5e8e974”,algorithm=MD5
Content-Length: 201

v=0
o=Zoiper_user 0 0 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (13 headers 10 lines) —
Sending to 10.10.0.188 : 5060 (NAT)
Using INVITE request as basis request - ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Found user ‘test’ for 'test’
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.10.0.188:8000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.0.188:8000
Looking for 100 in default (domain 10.10.0.157)
list_route: hop: sip:test@10.10.0.188:5060;transport=UDP

<— Transmitting (no NAT) to 10.10.0.188:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:100@10.10.0.157
Content-Length: 0

<------------>
– Executing [100@default:1] ReceiveFAX(“SIP/test-09fc2f08”, “/tmp/test.tiff”) in new stack
Audio is at 10.10.0.157 port 18112
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.10.0.188:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:100@10.10.0.157
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1718651154 1718651154 IN IP4 10.10.0.157
s=Asterisk PBX 1.6.0.9
c=IN IP4 10.10.0.157
t=0 0
m=audio 18112 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=sendrecv

<------------>
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
ACK sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-390d871a12b57aca-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:test@10.10.0.188:5060;transport=UDP
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 ACK
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username=“test”,realm=“asterisk”,nonce=“2b2a44fc”,uri="sip:100@10.10.0.157;transport=UDP",response=“977211f
ce0fadb5e8e974”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
INVITE sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:test@10.10.0.188:5060;transport=UDP
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username=“test”,realm=“asterisk”,nonce=“2b2a44fc”,uri="sip:100@10.10.0.157",response=“b9afeb630dfb17e475162
”,algorithm=MD5
Content-Length: 367

v=0
o=Zoiper_user 517578557 184362255 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=image 8000 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxVersion:0
a=T38FaxMaxBuffer:400
a=T38FaxTranscodingMMR:0
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxFillBitRemoval:0
a=T38MaxBitRate:14400
a=T38FaxMaxDatagram:400
a=T38FaxTranscodingJBIG:0

<------------->
— (13 headers 15 lines) —
Sending to 10.10.0.188 : 5060 (NAT)
Got T.38 offer in SDP in dialog ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)

<— Transmitting (NAT) to 10.10.0.188:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:100@10.10.0.157
Content-Length: 0

tank*CLI>
<— Reliably Transmitting (NAT) to 10.10.0.188:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
ACK sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1—d8754z-;rport
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
tank*CLI>
<— SIP read from UDP://10.10.0.188:5060 —>
BYE sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-f3bc204a253cbd65-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:test@10.10.0.188:5060;transport=UDP
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 4 BYE
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username=“test”,realm=“asterisk”,nonce=“2b2a44fc”,uri="sip:100@10.10.0.157",response=“a29fcc93c2b2e9fda666a
”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.10.0.188 : 5060 (NAT)

<— Transmitting (NAT) to 10.10.0.188:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-f3bc204a253cbd65-1—d8754z-;received=10.10.0.188;rport=5060
From: sip:test@10.10.0.157;transport=UDP;tag=765ae221
To: sip:100@10.10.0.157;transport=UDP;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 4 BYE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
[May 7 22:56:56] WARNING[29365]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.
[May 7 22:56:56] WARNING[29365]: app_fax.c:650 transmit: Transmission error
== Spawn extension (default, 100, 1) exited non-zero on 'SIP/test-09fc2f08’
Really destroying SIP dialog ‘ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.’ Method: BYE
tank*CLI>
[/size]
Thanks for any comments