asterisk users - Apr 2011 - 488 error in T38 Gatewaying in Asterisk 1.8 with patch 13405

Hello List,

I have been trying to setup T38 gatewaying with the following setup

SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO

I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo
Canceller: HWEC

I'm aware there's no support for T38 gateway but I have been trying to
get
the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems
like some people have been able to get transparent T38 gateway to work on
1.8

I have downloaded the latest patch
asterisk-1.8.4_fax.patch<https://issues.asterisk.org/file_download.php?file_id=28947&type=bug>
applied
it. Installed my spandsp spandsp-0.0.6pre18.tgz and it compiles succesfully.

I'm using zoiper to test the T38 faxing. Here is my dialplan

  '_5062XXXXXXX' => 1. Set(FAXOPT(t38gateway)=yes)
 [pbx_config]
                    2. Dial(DAHDI/g0/${EXTEN})
 [pbx_config]
                    3. Hangup()
[pbx_config]

  == Using SIP RTP CoS mark 5
    -- Executing [50624309954 at termination-test:1]
Set("SIP/robert-00000007",
"GROUP(customers)=ibasis") in new stack
    -- Executing [50624309954 at termination-test:2]
Set("SIP/robert-00000007",
"GROUP(termination)=ss7") in new stack
    -- Executing [50624309954 at termination-test:3]
Set("SIP/robert-00000007",
"FAXOPT(t38gateway)=yes") in new stack
    -- Executing [50624309954 at termination-test:4]
Dial("SIP/robert-00000007", "DAHDI/g0/50624309954") in new
stack
    -- Called g0/50624309954
    -- DAHDI/1-1 is proceeding passing it to SIP/robert-00000007
[Apr  8 00:26:35] NOTICE[31251]: chan_sip.c:8615 process_sdp: T.38 re-INVITE
detected but no fax extension
    -- Running Gateway activestate=4 (SIP/robert-00000007) and
inactivestate=0 (DAHDI/1-1)
[Apr  8 00:26:35] ERROR[10783]: res_fax.c:822 fax_session_reserve: Could not
locate a FAX technology module with capabilities (T38_GATEWAY)
[Apr  8 00:26:35] ERROR[10783]: res_fax.c:2527
__ast_t38_gateway_handle_parameters: Unable to reserve FAX session.
    -- Running Gateway activestate=0 (DAHDI/1-1) and inactivestate=4
(SIP/robert-00000007)
[Apr  8 00:26:35] ERROR[10783]: res_fax.c:822 fax_session_reserve: Could not
locate a FAX technology module with capabilities (T38_GATEWAY)
[Apr  8 00:26:35] ERROR[10783]: res_fax.c:2527
__ast_t38_gateway_handle_parameters: Unable to reserve FAX session.

The call gets established but the fax on the other side, receive nothing
and disconnect. The reciving fax complains about timeout.

I added the faxdetect app created by the patch. I asked the reported about
what it actually does as we don't know anything further than is related to
the T38 kickover

  '_5062XXXXXXX' => 1. Set(FAXOPT(t38gateway)=yes)
 [pbx_config]
                    2. FaxDetect(5)
[pbx_config]
                    3. Dial(DAHDI/g0/${EXTEN})
 [pbx_config]
                    4. Hangup()
[pbx_config]

  == Using SIP RTP CoS mark 5
    -- Executing [50624309954 at termination-test:1]
Set("SIP/robert-00000005",
"GROUP(customers)=ibasis") in new stack
    -- Executing [50624309954 at termination-test:2]
Set("SIP/robert-00000005",
"GROUP(termination)=ss7") in new stack
    -- Executing [50624309954 at termination-test:3]
Set("SIP/robert-00000005",
"FAXOPT(t38gateway)=yes") in new stack
    -- Executing [50624309954 at termination-test:4]
FaxDetect("SIP/robert-00000005", "5") in new stack
    -- Executing [50624309954 at termination-test:5]
Dial("SIP/robert-00000005", "DAHDI/g0/50624309954") in new
stack
    -- Called g0/50624309954
    -- DAHDI/1-1 is proceeding passing it to SIP/robert-00000005
    -- DAHDI/1-1 answered SIP/robert-00000005

But then the call drop after 5 seconds with an 488 not acceptable here.

o=ZoiperCommunicator_user 0 1 IN IP4 192.168.1.60
s=ZoiperCommunicator_session
c=IN IP4 192.168.1.60
t=0 0
m=image 8000 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

Asterisk returns

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.60:53325
;branch=z9hG4bK-d8754z-35a890bc09845fb2-1---d8754z-;received=201.192.28.178;rport=53325
From: "robert"<sip:robert at
190.106.66.210;transport=UDP>;tag=0905b421
To: <sip:50624309954 at 190.106.66.210;transport=UDP>;tag=as605221e3
Call-ID: NGU5YzQ5NDdiNTU3ZDg1N2VmYTM3MGQwNDhlMjlkNGE.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

I can see this type of behaviour with T38 is quite common

https://issues.asterisk.org/view.php?id=16327
https://issues.asterisk.org/view.php?id=16793
https://issues.asterisk.org/view.php?id=16793

I was wondering if anyone has some experience with this patch, or T38
asterisk transparent gateway.

I think the problem could be more on the sip channel rejecting the call, can
someone help me try to narrow down the issue. I have more captures and logs
available

-- 
Robert
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