[asterisk-bugs] [Asterisk 0013405]: [patch] T38 gateway

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13405 
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Reported By:                dafe_von_cetin
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13405
Category:                   Applications/app_fax
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 140548 
Request Review:              
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Date Submitted:             2008-08-30 16:44 CDT
Last Modified:              2009-04-29 13:29 CDT
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Summary:                    [patch] T38 gateway
Description: 
Hi all,

I'm sending you patch containing new application app_faxgateway.c
("FaxGateway") which is able to mediate T30 to T38 and vice versa.
Feature is using spands library (I used spandsp-0.0.4pre18 and
spandsp-0.0.5pre4).

Best regards
Daniel.

====================================================================== 

---------------------------------------------------------------------- 
 (0103955) gentian (reporter) - 2009-04-29 13:29
 http://bugs.digium.com/view.php?id=13405#c103955 
---------------------------------------------------------------------- 
I tried Asterisk 1.6.0.9 with t38-gw-r154965.patch, compile is ok, no
errors. But when the call goes to ATA Linksys SPA2102 or Grandstream H286
or Zhone DSLAM with POTS ports the call fails. Every time the call is
connected and switched to T38 all clients respond with:

SIP/2.0 488 Not acceptable here 

and sends HangupCause: Normal Clearing so DAHDI channel hungs up and call
is finished failing the fax.

some of the logs:

yy.yy.yy.yy = Asterisk Server
xx.xx.xx.xx = Linksys SPA2102

zzzzzzzzz = SIP number register on SPA to Asterisk server
nnnnnnnnn = PSTN number sending the FAX



<--- SIP read from UDP://xx.xx.xx.xx:5060 --->
INVITE sip:zzzzzzzzz at yy.yy.yy.yy SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-d228ac1c
From: <sip:zzzzzzzzz at xx.xx.xx.xx:5060>;tag=859e1fcb4af7a47bi0
To: "nnnnnnnnn" <sip:zzzzzzzzz at yy.yy.yy.yy>;tag=as72e031a7
Remote-Party-ID: zzzzzzzzz
<sip:zzzzzzzzz at yy.yy.yy.yy>;screen=yes;party=called
Call-ID: 6b1216784093afb863cb5a614ce26d45 at yy.yy.yy.yy
CSeq: 101 INVITE
Max-Forwards: 70
Contact: zzzzzzzzz <sip:zzzzzzzzz at xx.xx.xx.xx:5060>
Expires: 30
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 265
Content-Type: application/sdp

v=0
o=- 35420 35420 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=image 16466 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy


<------------->
--- (13 headers 12 lines) ---
Sending to xx.xx.xx.xx : 5060 (no NAT)
Got T.38 offer in SDP in dialog
6b1216784093afb863cb5a614ce26d45 at yy.yy.yy.yy
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
Callid 6b1216784093afb863cb5a614ce26d45 at yy.yy.yy.yy
Capabilities: us - 0x8 (alaw), 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 xx.xx.xx.xx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bK-d228ac1c;received=xx.xx.xx.xx
From: <sip:zzzzzzzzz at xx.xx.xx.xx:5060>;tag=859e1fcb4af7a47bi0
To: "nnnnnnnnn" <sip:zzzzzzzzz at yy.yy.yy.yy>;tag=as72e031a7
Call-ID: 6b1216784093afb863cb5a614ce26d45 at yy.yy.yy.yy
CSeq: 101 INVITE
User-Agent: AsteriskPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:zzzzzzzzz at yy.yy.yy.yy>
Content-Length: 0


<------------>
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
s1*CLI>
<--- Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bK-d228ac1c;received=xx.xx.xx.xx
From: <sip:zzzzzzzzz at xx.xx.xx.xx:5060>;tag=859e1fcb4af7a47bi0
To: "nnnnnnnnn" <sip:zzzzzzzzz at yy.yy.yy.yy>;tag=as72e031a7
Call-ID: 6b1216784093afb863cb5a614ce26d45 at yy.yy.yy.yy
CSeq: 101 INVITE
User-Agent: AsteriskPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-29 13:29 gentian        Note Added: 0103955                          
======================================================================