[asterisk-bugs] [Asterisk 0014769]: [patch] Improvements/fixes for app_fax

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14769 
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Reported By:                andrew
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14769
Category:                   Applications/app_fax
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for review
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 184382 
Request Review:              
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Date Submitted:             2009-03-26 15:14 CDT
Last Modified:              2009-04-07 03:05 CDT
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Summary:                    [patch] Improvements/fixes for app_fax
Description: 

I was working on a backport of app_fax to asterisk 1.4 branch.

It works great now, but the original code was causing errors to be logged
(a new issue as of 1.4.24, see bug id http://bugs.digium.com/view.php?id=14723).
As I looked into the code
more I found what I think a problems/bugs. Here is a patch (for 1.6 trunk)
for my "improvements/fixes".

I'm using SPANDSP 0.0.6pre7 and testing with SIP using an external PSTN
gateway (AS5300). I can send a fax from the system to the same system
(using the PSTN), so it seems happy. I also tested with a normal fax
machine. RX works way better than TX for me.

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Relationships       ID      Summary
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related to          0014812 ast_read() used with incorrect ast_wait...
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---------------------------------------------------------------------- 
 (0102814) sgimeno (reporter) - 2009-04-07 03:05
 http://bugs.digium.com/view.php?id=14769#c102814 
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I don't think Asterisk should offer Cisco T.38 if the call was initiated in
audio mode, but the other way around. It's the callee (Cisco) that should
offer T.38 to the caller (Asterisk).

The call flow should be:

      Asterisk         Cisco        PSTN-FAX
          |              |              |
          |              |              |
          |              |              |
          |Invite (Audio)|              |
          |------------->|              |
          |200 OK (Audio)|              |
          |<-------------|              |
          |ACK           |              |
          |------------->|              |
          |              |Establishes the call with the PSTN Fax
          |              |              |
          |re-INVITE (T.38)             |
          |<-------------|              |
          |200 OK (T.38) |              |
          |------------->|              |
          |ACK           |              |
          |<-------------|              |
          |              |              |
          |              |              |

The problem I'm having is that Asterisk answers with a "488 Not acceptable
here", so the Cisco downgrades the call to audio again. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-07 03:05 sgimeno        Note Added: 0102814                          
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