[asterisk-bugs] [Asterisk 0016327]: Asterisk responds 488 - Not acceptable here on T38 reinvite

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16327 
====================================================================== 
Reported By:                serje
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16327
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           Older 1.6.1 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-11-26 03:48 CST
Last Modified:              2009-12-03 16:35 CST
====================================================================== 
Summary:                    Asterisk responds 488 - Not acceptable here on T38
reinvite
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.
====================================================================== 

---------------------------------------------------------------------- 
 (0114690) Hubguru (reporter) - 2009-12-03 16:35
 https://issues.asterisk.org/view.php?id=16327#c114690 
---------------------------------------------------------------------- 
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 at 208.81.54.18 SIP/2.0
To: "Ntegrated Solutions" <sip:2142698390 at 208.81.54.18>;tag=as17699b02
From: <sip:1012144466744 at 208.81.54.27>;tag=67c0330c-25793f65-1b3651d0
Call-ID: 731f9b4066839c95744eda0317e3c882 at 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 at 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 at 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 at 208.81.54.27>;tag=67c0330c-25793f65-1b3651d0
To: "Ntegrated Solutions" <sip:2142698390 at 208.81.54.18>;tag=as17699b02
Call-ID: 731f9b4066839c95744eda0317e3c882 at 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 at 208.81.54.18>
Content-Length: 0 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-03 16:35 Hubguru        Note Added: 0114690                          
======================================================================