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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=13405 
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Reported By:                dafe_von_cetin
Assigned To:                
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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:              
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Date Submitted:             2008-08-30 16:44 CDT
Last Modified:              2011-01-07 17:06 CST
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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.

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 (0130322) dluzin (reporter) - 2011-01-07 17:06
 https://issues.asterisk.org/view.php?id=13405#c130322 
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I meant Asterisk 1.6.2.13 No 1 and Asterisk 1.6.2.13 No 2 run on the same
machine bind to 5061 sip port.
My setup works as following:

Extension on Asterisk No 1 (in custom context) initiates call to some
number.
Custom context changes number (add leading '1' to the number) and sends
number for regular routing.
There is a route for numbers beginning from '1' that sends this call to
Asterisk No 2 using regular Dial command (with rtTwW options). 
Asterisk No 2 strips leading '1' from the number and sends it back to
Asterisk No 1 using t.38 Dial (with b option).
Asterisk No 1 sends received number (same as initially dialed by
extension) for regular routing and makes outgoing call to one of upstream
Asterisk 1.4.24.1 using regular Dial command.

At the end I have full Dial function functionality and t.38 gateway as
well as your favorite web-interface for trunks, peers, extensions, etc.
This setup is quite complicated, but tuned once works for years.
I had similar setup of Asterisk 1.4.24.1 and callweaver 1.2 for about 1.5
years.

Regarding your problem - check "canreinvite" parameters of all peers
within ata1 <-> ata2 path. Try putting "canreinvite=no" to each peer and
"nat=yes" to prevent changing of RTP/UDPTL media path and engage UDPTL
passthrough.
Put to sip.conf t38pt_udptl=yes, redundancy,maxdatagram=400 and udptlstart
and udptlend to udptl.conf
Do not put t38pt_usertpsource - in my case it was causing loss of udptl
packets (why? I don't know).
Spandsp's modem on my x86_64 was not working properly (no sound at t.30
side), so I had to upgrade spandsp library from 0.0.5_pre4 to 0.0.6_pre12.
On x86 0.0.5_pre4 works well.
If you will upgrade spandsp do not forget to reconfigure and recompile
asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-07 17:06 dluzin         Note Added: 0130322                          
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