[asterisk-bugs] [Asterisk 0016795]: 1.4 does not send any SIP messages after the "100 Trying" to the T.38 INVITE requesting side

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16795 
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Reported By:                vrban
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16795
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.4.29 
JIRA:                       SWP-891 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-02-09 11:07 CST
Last Modified:              2010-02-11 05:34 CST
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Summary:                    1.4 does not send any SIP messages after the "100
Trying" to the T.38 INVITE requesting side
Description: 

Asterisk 1.4 does not send any SIP messages after the "100 Trying" to the
T.38 INVITE requesting side. chan_sip talk to the B-side, where itself send
out the T.38 re-INVITE, but does not send anything to the originating
A-side.

"sip show channels" show this dead A side channels which never get closed:

111.222.111.222  0123456789  69570b5c142b447  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  03f0bc4832e47e8  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  10a623dc4650f90  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  5636aa6171efef3  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  5877536469f446e  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  3a83a6480cf7eed  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  664d8ac26547274  0x0 (nothing)    No      
Tx: INVITE
111.222.111.222  0123456789  38bb8ada68b44ae  0x0 (nothing)    No      
Tx: INVITE

With 1.6 and trunk chan_sip is talking to A-side.

In the attached sip trace, you see the only answer to C.C.C.C is only the
100 Trying, then nothing.
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 (0117984) vrban (reporter) - 2010-02-11 05:33
 https://issues.asterisk.org/view.php?id=16795#c117984 
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I think it's much easier to extend the sip_handle_t38_reinvite in 1.4
chan_sip
that it handel also 488 for outgoing T.38 re-INVITE, and forwar the 488 to
the originating party.

To test this, i just harcoded a:
transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
into sip_handle_t38_reinvite and a bit more also in
handle_response_invite, where the 488 case is. then chan_sip can handle
also the T.38 to audio fallback.

Is this the correct way to handel this? Then i try to make a clean patch
of it. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-11 05:34 vrban          Note Added: 0117984                          
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