[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 noreply at bugs.digium.com
Tue Jun 1 09:54:06 CDT 2010


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:              2010-06-01 09:54 CDT
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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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Relationships       ID      Summary
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related to          0016692 [patch] Missing fallback to audio fax f...
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---------------------------------------------------------------------- 
 (0122687) svnbot (reporter) - 2010-06-01 09:54
 https://issues.asterisk.org/view.php?id=16795#c122687 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 266579

U   branches/1.4/channels/chan_sip.c

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r266579 | pabelanger | 2010-06-01 09:54:05 -0500 (Tue, 01 Jun 2010) | 18
lines

Missing fallback to audio fax feature when T.38 re-INVITE failed

When a T.38 re-INVITE failed with an 488 or 606 answer, we should
fallback to audio fax by send a re-re-INVITE without T.38. The
function is backported from 1.6 asterisk.

(closes issue https://issues.asterisk.org/view.php?id=16795)
Reported by: vrban

(closes issue https://issues.asterisk.org/view.php?id=16692)
Reported by: vrban
Patches:
      t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
Tested by: lmadsen, vrban, haggard

https://reviewboard.asterisk.org/r/514/


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http://svn.digium.com/view/asterisk?view=rev&revision=266579 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-01 09:54 svnbot         Checkin                                      
2010-06-01 09:54 svnbot         Note Added: 0122687                          
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