ASTERISK-15218: Asterisk responds 488 - Not acceptable here on T38 reinvite

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Summary: ASTERISK-15218: Asterisk responds 488 - Not acceptable here on T38 reinvite

Reporter: Sergei (serje) Labels:

Date Opened: 2009-11-26 03:48:11.000-0600 Date Closed: 2011-07-26 14:44:21

Priority: Major Regression? No

Status: Closed/Complete Components: Channels/chan_sip/T.38

Versions: Frequency of
Occurrence

Related
Issues:

Environment: Attachments: ( 0) Asterisk_1.6_fax.zip

Description: I'm trying to send a fax from asterisk through Audiocodes Mediant1000 gateway to PSTN using T38.
The call is being set up in audio, then Mediant sends reinvite with T38 - and asterisk responds "488 Not acceptable here".
This can be observed in 1.6.0.15, 1.6.1.5, 1.6.1.9 (current 1.6 trunk and latest releases nave another bug (is is on bug tracker) - they can't establish connection on sip trunk, so I couldn't verify, whether T38 issue present in them.

****** ADDITIONAL INFORMATION ******

astfax*CLI>
-- Attempting call on SIP/trunk_1/992378471 for 1000@DLPN_City_call:1 (Retry 1)
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Audio is at 172.17.18.135 port 10828
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.17.15.7:5060:
INVITE sip:992378471@172.17.15.7 SIP/2.0
Via: SIP/2.0/UDP 172.17.18.135:5060;branch=z9hG4bK1a47bbf5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
To: <sip:992378471@172.17.15.7>
Contact: <sip:asterisk@172.17.18.135>
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.6
Date: Thu, 26 Nov 2009 09:34:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 651922310 651922310 IN IP4 172.17.18.135
s=Asterisk PBX 1.6.1.6
c=IN IP4 172.17.18.135
t=0 0
m=audio 10828 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
astfax*CLI>
<--- SIP read from UDP://172.17.15.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.18.135:5060;branch=z9hG4bK1a47bbf5;rport
From: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
To: <sip:992378471@172.17.15.7>;tag=1c1233090922
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
astfax*CLI>
<--- SIP read from UDP://172.17.15.7:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.17.18.135:5060;branch=z9hG4bK1a47bbf5;rport
From: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
To: <sip:992378471@172.17.15.7>;tag=1c1233090922
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 102 INVITE
Contact: <sip:1008@172.17.15.7:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Content-Type: application/sdp
Content-Length: 253

v=0
o=AudiocodesGW 1233180816 1233180484 IN IP4 172.17.15.7
s=Phone-Call
c=IN IP4 172.17.15.7
t=0 0
m=audio 6080 RTP/AVP 0 101
c=IN IP4 172.17.15.7
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.17.15.7:6080
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 172.17.15.7:6080

<--- SIP read from UDP://172.17.15.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.18.135:5060;branch=z9hG4bK1a47bbf5;rport
From: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
To: <sip:992378471@172.17.15.7>;tag=1c1233090922
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 102 INVITE
Contact: <sip:1008@172.17.15.7:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Content-Type: application/sdp
Content-Length: 253

v=0
o=AudiocodesGW 1233180816 1233180484 IN IP4 172.17.15.7
s=Phone-Call
c=IN IP4 172.17.15.7
t=0 0
m=audio 6080 RTP/AVP 0 101
c=IN IP4 172.17.15.7
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
list_route: hop: <sip:1008@172.17.15.7:5060>
set_destination: Parsing <sip:1008@172.17.15.7:5060> for address/port to send to
set_destination: set destination to 172.17.15.7, port 5060
Transmitting (no NAT) to 172.17.15.7:5060:
ACK sip:1008@172.17.15.7:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.18.135:5060;branch=z9hG4bK1b9bcabe;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
To: <sip:992378471@172.17.15.7>;tag=1c1233090922
Contact: <sip:asterisk@172.17.18.135>
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.6
Content-Length: 0

---
> Channel SIP/trunk_1-09cbf0b8 was answered.
-- Executing [1000@DLPN_City_call:1] Answer("SIP/trunk_1-09cbf0b8", "") in new stack
-- Executing [1000@DLPN_City_call:2] Wait("SIP/trunk_1-09cbf0b8", "2") in new stack
-- Executing [1000@DLPN_City_call:3] Wait("SIP/trunk_1-09cbf0b8", "2") in new stack
astfax*CLI>
<--- SIP read from UDP://172.17.15.7:5060 --->
INVITE sip:asterisk@172.17.18.135 SIP/2.0
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1503317125
Max-Forwards: 70
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 1 INVITE
Contact: <sip:1008@172.17.15.7:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Content-Type: application/sdp
Content-Length: 313

v=0
o=AudiocodesGW 1233180816 1233180485 IN IP4 172.17.15.7
s=Phone-Call
c=IN IP4 172.17.15.7
t=0 0
m=image 6082 udptl t38
c=IN IP4 172.17.15.7
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (13 headers 13 lines) ---
Sending to 172.17.15.7 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 74cfd94161789ddc786011045b5b0df6@172.17.18.135
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 74cfd94161789ddc786011045b5b0df6@172.17.18.135
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 (no NAT) to 172.17.15.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1503317125;received=172.17.15.7
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:asterisk@172.17.18.135>
Content-Length: 0

<------------>
-- Executing [1000@DLPN_City_call:4] SendFAX("SIP/trunk_1-09cbf0b8", "$FAXDIR/$FILENAME") in new stack
astfax*CLI>
<--- Reliably Transmitting (no NAT) to 172.17.15.7:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1503317125;received=172.17.15.7
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

<------------>
astfax*CLI>
<--- SIP read from UDP://172.17.15.7:5060 --->
ACK sip:asterisk@172.17.18.135 SIP/2.0
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1503317125
Max-Forwards: 70
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 1 ACK
Contact: <sip:1008@172.17.15.7:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Content-Length: 0

astfax*CLI>
<--- SIP read from UDP://172.17.15.7:5060 --->
BYE sip:asterisk@172.17.18.135 SIP/2.0
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1988441121
Max-Forwards: 70
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 2 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.030.003
Reason: Q.850 ;cause=17 ;text="local"
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 172.17.15.7 : 5060 (no NAT)

<--- Transmitting (no NAT) to 172.17.15.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.15.7;branch=z9hG4bKac1988441121;received=172.17.15.7
From: <sip:992378471@172.17.15.7>;tag=1c1233090922
To: "asterisk" <sip:asterisk@172.17.18.135>;tag=as58653775
Call-ID: 74cfd94161789ddc786011045b5b0df6@172.17.18.135
CSeq: 2 BYE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Nov 26 11:35:21] WARNING[7959]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.
[Nov 26 11:35:21] WARNING[7959]: app_fax.c:701 transmit: Transmission error
== Spawn extension (DLPN_City_call, 1000, 4) exited non-zero on 'SIP/trunk_1-09cbf0b8'
[Nov 26 11:35:21] NOTICE[7959]: pbx_spool.c:348 attempt_thread: Call completed to SIP/trunk_1/992378471
Really destroying SIP dialog '74cfd94161789ddc786011045b5b0df6@172.17.18.135' Method: BYE

Comments: By: linuxrulez (linuxrulez) 2009-12-03 03:59:00.000-0600

Confirming for 1.6.2r231696:

[Dec 3 11:39:52] -- Accepting AUTHENTICATED call from XXXXX:
> requested format = alaw,
> requested prefs = (alaw|ulaw|slin|gsm|g729),
> actual format = alaw,
> host prefs = (alaw|slin|gsm|g729),
> priority = mine
[Dec 3 11:39:52] -- Executing [test@from_voip:1] FaxGateway("IAX2/test-5099", "SIP/faxer/703,10,R") in new stack
[Dec 3 11:39:52] == Using SIP RTP CoS mark 5
[Dec 3 11:39:52] == Using UDPTL CoS mark 5
[Dec 3 11:39:52] Audio is at XXXX port 15636
[Dec 3 11:39:52] Adding codec 0x2 (gsm) to SDP
[Dec 3 11:39:52] Adding codec 0x4 (ulaw) to SDP
[Dec 3 11:39:52] Adding codec 0x8 (alaw) to SDP
[Dec 3 11:39:52] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 3 11:39:52] Reliably Transmitting (no NAT) to XXXXXX:5061:
INVITE sip:703@XXX:5061 SIP/2.0
Via: SIP/2.0/UDP XXXXX:5060;branch=z9hG4bK588ad1dd;rport
Max-Forwards: 70
From: "703" <sip:703@XXXX>;tag=as31a0cb3e
To: <sip:703@XXXXX:5061>
Contact: <sip:703@XXXXX>
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
Date: Thu, 03 Dec 2009 09:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 1106053413 1106053413 IN IP4 XXXXX
s=Asterisk PBX SVN-branch-1.6.2-r231696M
c=IN IP4 XXXXXX
t=0 0
m=audio 15636 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXXX:5061 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXXXXX
From: "703" <sip:703@XXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXX
To: <sip:703@XXXXX:5061>
Contact: <sip:703@XXXXX:5061>
Content-Length: 0

<------------->
[Dec 3 11:39:52] --- (8 headers 0 lines) ---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXX:5061 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXX
User-Agent: T38Modem/1.2.1
From: "703" <sip:703@XXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXX
Organization: Vyacheslav Frolov
To: <sip:703@XXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Length: 0

<------------->
[Dec 3 11:39:52] --- (10 headers 0 lines) ---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXXX:5061 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXXXXXX
User-Agent: T38Modem/1.2.1
From: "703" <sip:703@XXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXXXXXX
Organization: Vyacheslav Frolov
To: <sip:703@XXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Contact: <sip:703@XXXXXXx:5061>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 1259833192 1 IN IP4 XXXXXX
s=T38Modem/1.2.1
c=IN IP4 XXXXXX
t=0 0
m=audio 5002 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240

<------------->
[Dec 3 11:39:52] --- (12 headers 11 lines) ---
[Dec 3 11:39:52] Found RTP audio format 0
[Dec 3 11:39:52] Found RTP audio format 101
[Dec 3 11:39:52] Found audio description format PCMU for ID 0
[Dec 3 11:39:52] Found audio description format telephone-event for ID 101
[Dec 3 11:39:52] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Dec 3 11:39:52] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Dec 3 11:39:52] Peer audio RTP is at port XXXX:5002
[Dec 3 11:39:52] list_route: hop: <sip:703@XXXX:5061>
[Dec 3 11:39:52] set_destination: Parsing <sip:703@XXX:5061> for address/port to send to
[Dec 3 11:39:52] set_destination: set destination to XXXX, port 5061
[Dec 3 11:39:52] Transmitting (no NAT) to XXXX:5061:
ACK sip:703@XXXXx:5061 SIP/2.0
Via: SIP/2.0/UDP XXXXX:5060;branch=z9hG4bK3fa801fb;rport
Max-Forwards: 70
From: "703" <sip:703@XXXX>;tag=as31a0cb3e
To: <sip:703@XXXX:5061>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Contact: <sip:703@XXXXX>
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXX
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
Content-Length: 0

---
[Dec 3 11:39:53]
<--- SIP read from UDP:XXXX:5061 --->
INVITE sip:703@XXXXXXXX SIP/2.0
CSeq: 2 INVITE
Via: SIP/2.0/UDP XXXXXXX:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;rport
User-Agent: T38Modem/1.2.1
From: <sip:703@XXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXX
Organization: Vyacheslav Frolov
To: "703" <sip:703@XXXXXX>;tag=as31a0cb3e
Contact: <sip:703@XXXXX:5061>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 187
Max-Forwards: 70

v=0
o=- 1259833192 2 IN IP4 XXXXX
s=T38Modem/1.2.1
c=IN IP4 XXXXXXX
t=0 0
m=image 5002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38FaxRateManagement:transferredTCF

<------------->
[Dec 3 11:39:53] --- (13 headers 9 lines) ---
[Dec 3 11:39:53] Sending to XXXXXX : 5061 (no NAT)
[Dec 3 11:39:53] Got T.38 offer in SDP in dialog 4ddcdbca767a3f4842ad448c5a7c3321@XXXXXX
[Dec 3 11:39:53] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
[Dec 3 11:39:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 3 11:39:53] Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Dec 3 11:39:53] WARNING[22525]: udptl.c:766 calculate_far_max_ifp: (no tag): Cannot calculate far_max_ifp before far_max_datagram has been set.
[Dec 3 11:39:53]
<--- Transmitting (no NAT) to XXXXX:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXXXx:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;received=XXXXXx;rport=5061
From: <sip:703@XXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
To: "703" <sip:703@XXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXXX
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.6.2-r231696M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:703@XXXXX>
Content-Length: 0

<--- Reliably Transmitting (no NAT) to XXXXX:5061 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP XXXXXX:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;received=XXXXX;rport=5061
From: <sip:703@XXXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
To: "703" <sip:703@XXXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXx
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.6.2-r231696M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

<------------>
[Dec 3 11:39:58]
<--- SIP read from UDP:XXXXX:5061 --->
ACK sip:703@XXXXx SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP XXXXXx:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;rport
From: <sip:703@XXXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXXXX
To: "703" <sip:703@XXXXXx>;tag=as31a0cb3e
Content-Length: 0
Max-Forwards: 70

<------------->
[Dec 3 11:39:58] --- (8 headers 0 lines) ---

< ONLY SILENCE HERE, no FAX tones (SIP connected to T38modem, connected to FaxGetty from HylaFAX+>

[Dec 3 11:40:07] Scheduling destruction of SIP dialog '4ddcdbca767a3f4842ad448c5a7c3321@XXXXX' in 32000 ms (Method: ACK)
[Dec 3 11:40:07] set_destination: Parsing <sip:703@XXXX:5061> for address/port to send to
[Dec 3 11:40:07] set_destination: set destination to XXXXx, port 5061
[Dec 3 11:40:07] Reliably Transmitting (no NAT) to XXXX:5061:
BYE sip:703@XXXXXxx:5061 SIP/2.0
Via: SIP/2.0/UDP XXXx:5060;branch=z9hG4bK01f98963;rport
Max-Forwards: 70
From: "703" <sip:703@XXXXXX>;tag=as31a0cb3e
To: <sip:703@XXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXX
CSeq: 103 BYE
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
[Dec 3 11:40:07] -- Hungup 'IAX2/test-5099'
[Dec 3 11:40:07]
<--- SIP read from UDP:XXXXx:5061 --->
SIP/2.0 200 OK
CSeq: 103 BYE
Via: SIP/2.0/UDP XXXXXx:5060;branch=z9hG4bK01f98963;rport
From: "703" <sip:703@XXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321@XXXXXx
To: <sip:703@XXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Content-Length: 0

<------------->
[Dec 3 11:40:07] --- (7 headers 0 lines) ---
[Dec 3 11:40:07] SIP Response message for INCOMING dialog BYE arrived
[Dec 3 11:40:07] Really destroying SIP dialog '4ddcdbca767a3f4842ad448c5a7c3321@XXXXX' Method: ACK
By: JR Richardson (hubguru) 2009-12-03 16:35:00.000-0600

I'm getting similar results trying to get t38 established to a Lucent MAX TNT version 14.03, using Asterisk 1.6.1.10 and 3 different ATA's. My setup:
ATA>NAT<>Asterisk>Pulbic<>MAX TNT><PRI

Here are the SIP debug:

Normal call setup ulaw, then once fax tone hits the line, the TNT sends an invite to Asterisk with the proper T38 messaging, Asterisk responds with: "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session."

<--- SIP read from UDP://208.81.54.27:5060 --->
INVITE sip:2142698390@208.81.54.18 SIP/2.0
To: "Ntegrated Solutions" <sip:2142698390@208.81.54.18>;tag=as17699b02
From: <sip:1012144466744@208.81.54.27>;tag=67c0330c-25793f65-1b3651d0
Call-ID: 731f9b4066839c95744eda0317e3c882@208.81.54.18
CSeq: 7480 INVITE
Via: SIP/2.0/UDP 208.81.54.27:5060;branch=z9hG4bK00012c597faf2ef1
Max-Forwards: 70
Contact: <sip:2142698390@208.81.54.27:5060;user=phone>
Content-Type: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
Content-Length: 364

v=0
o=t1gw01 628703077 628703078 IN IP4 208.81.54.27
s=Session SDP
c=IN IP4 208.81.54.27
t=0 0
m=image 32764 udptl t38
a=T38FaxMaxDatagram:316
a=T38FaxMaxBuffer:72
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxVersion:0
a=T38FaxTranscodingJBIG:0
a=T38FaxTranscodingMMR:0
a=T38FaxFillBitRemoval:0
a=T38MaxBitRate:14400

<------------->
--- (14 headers 15 lines) ---
Sending to 208.81.54.27 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 731f9b4066839c95744eda0317e3c882@208.81.54.18
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)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<--- Transmitting (no NAT) to 208.81.54.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.81.54.27:5060;branch=z9hG4bK00012c597faf2ef1;received=208.81.54.27
From: <sip:1012144466744@208.81.54.27>;tag=67c0330c-25793f65-1b3651d0
To: "Ntegrated Solutions" <sip:2142698390@208.81.54.18>;tag=as17699b02
Call-ID: 731f9b4066839c95744eda0317e3c882@208.81.54.18
CSeq: 7480 INVITE
Server: Asterisk PBX 1.6.1.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:2142698390@208.81.54.18>
Content-Length: 0
By: Sergei (serje) 2010-02-01 07:14:34.000-0600

Hi, are there any ideas - will it be fixed, or some workaround exists?

Thank you.
By: Max Molchanov (censo) 2010-02-25 12:36:06.000-0600

This looks to happen in chan_sip.c, handle_request_invite() around line 19547 in trunk. When re-invite is received, "100 trying" is sent out and 5000ms timer is started to send "488 not acceptable here" reply. Since the other party sends nothing in reply to "100 trying" and we do not send anything as well, call is terminated in 5 seconds.

case AST_STATE_UP:
ast_debug(2, "%s: This call is UP.... \n", c->name);
transmit_response(p, "100 Trying", req);

if (p->t38.state == T38_PEER_REINVITE) {
p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, dialog_ref(p, "passing dialog
} else if (p->t38.state == T38_ENABLED) {
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->
} else if (p->t38.state == T38_DISABLED) {

We are waiting for something from other party? But it looks like they should not send anything else.

By: Matthew Nicholson (mnicholson) 2011-07-26 14:43:26.532-0500

This occurs because SendFAX is executed after the t38 reinvite is received. This should be fixed in the latest res_fax and asterisk releases.
By: Matthew Nicholson (mnicholson) 2011-07-26 14:43:50.328-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions