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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Mar 25 05:42:07 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=13405 
====================================================================== 
Reported By:                dafe_von_cetin
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13405
Category:                   Applications/app_fax
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                       SWP-115 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/459/ 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 140548 
Request Review:              
====================================================================== 
Date Submitted:             2008-08-30 16:44 CDT
Last Modified:              2010-03-25 05:41 CDT
====================================================================== 
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.

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

---------------------------------------------------------------------- 
 (0119870) klaus3000 (reporter) - 2010-03-25 05:41
 https://issues.asterisk.org/view.php?id=13405#c119870 
---------------------------------------------------------------------- 
@dalepbx: 
1. in your setup, if both ATAs support T.38, then the t38-gateway will not
be activated as there is no need for it.
2. in your setup, Asterisk offloads RTP by sending reINVITEs, so that the
RTP stream will be directly between the ATAs. to prevent the reINVITEs set
canreinvite=no in sip.conf 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-25 05:41 klaus3000      Note Added: 0119870                          
======================================================================




More information about the asterisk-bugs mailing list