[asterisk-bugs] [Asterisk 0008677]: Asterisk does not reinvite peer for G.711 after T.38 negotiated failed with a "488" Event

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Mar 10 14:56:30 CDT 2008


The following issue has been ASSIGNED. 
====================================================================== 
http://bugs.digium.com/view.php?id=8677 
====================================================================== 
Reported By:                alex-911
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   8677
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4 
SVN Revision (number only!): 59083 
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             12-27-2006 08:13 CST
Last Modified:              03-10-2008 14:56 CDT
====================================================================== 
Summary:                    Asterisk does not reinvite peer for G.711 after T.38
negotiated failed with a "488" Event
Description: 
I have a linksys ATA connected to asterisk, configured for G.711 fax
passthru. asterisk is connected to a Cisco PSTN gateway. the default fax
protocol of the PSTN gateway is T.38, so if the Cisco media gateway detects
faxtone, there is a reinvite for T.38.
asterisk passes the reinvite down to the ATA. the ATA answers correctly
with a "488 not acceptable here".
instead of passing the 488 up to the proxy, asterisk seems to stop here.
it receives the retransmit of the reinvite and answers with a "503
Unavailable".
I would expect asterisk to pass the 488 up what would trigger another
reinvite for T.30 fax (G.711 passthru).

I'll post the simple call flow and the console log below. let me know if
more details are required.
10.10.10.23: ATA
172.16.16.111: *
172.16.16.155: SIP Proxy
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0009345 Problems
====================================================================== 

---------------------------------------------------------------------- 
 svnbot - 03-10-08 14:56  
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 107157

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r107157 | file | 2008-03-10 14:56:24 -0500 (Mon, 10 Mar 2008) | 4 lines

If we receive a 488 on a T38 request reinvite back to audio. As well
reinvite across a bridge back to audio if one side doesn't negotiate to
T38.
(closes issue http://bugs.digium.com/view.php?id=8677)
Reported by: alex-911

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=107157 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
03-10-08 14:56  svnbot         Note Added: 0083685                          
03-10-08 14:56  svnbot         Status                   acknowledged => assigned
======================================================================




More information about the asterisk-bugs mailing list