[asterisk-bugs] [Asterisk 0010274]: T.38 passthrough with 2 Asterisk boxes not working
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jul 30 13:11:34 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10274
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Reported By: cstadlmann
Assigned To:
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Project: Asterisk
Issue ID: 10274
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: block
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 76410
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 07-23-2007 08:41 CDT
Last Modified: 07-30-2007 13:11 CDT
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Summary: T.38 passthrough with 2 Asterisk boxes not working
Description:
This is the configuration I'm using:
ATA 1 <-> Asterisk 1 <-> Asterisk 2 <-> ATA 2
As long ATA's are resided on the same Asterisk box, no T.38 problem. But
if ATA 1 is trying to send a fax via T.38 to ATA 2, the complete call fails
because Asterisk 1 is killing it during re-invite and Asterisk 2 gives up.
Both Asterisk boxes are: SVN-branch-1.4-r76410
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Relationships ID Summary
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has duplicate 0010317 Bug 10274 seems to be SIP related, not ...
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cstadlmann - 07-30-07 13:11
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Puh... I really thought noone concerns about that issue...
To 1)
During the last week's testing and tracking and tracing I found out that
every ATA or other device immediately sends a '100 Trying' after it
receives the reinvite. After that the device responds with '200 OK'. So if
Asterisk1 would also send a 100 response immedately to Asterisk2 I think
the whole situation wouldn't go that far in the wrong direction. I tried to
figure out where to put in that code snippet, but had no luck because I'm
not familiar with chan_sip.c at all.
To 2)
As I mentioned before, Asterisk1 sends a completely senseless reinvite to
ATA1 with NO T.38 information, just plain RTP information. Please have a
look at line 3424 of 'Log Asterisk 1.txt' and below. You will see what I
mean. While ATA1 is responding with a '200 OK' for the T.38 reinvite it
receives that useless invite , reponds with '200 OK' again and 'falls back'
to plain RTP, so Asterisk 1 thinks that ATA 1 is not capable of T.38. This
is that I meant with Asterisk is not handling '100 Trying' correctly
because it sends wrong invite packets.
So again, I think that all problems are located in the '100 Trying'
message. Maybe you can implement that code segment into chan_sip.c or just
tell me where I can find it and I will try to patch by myself although I'm
not a C guru... Anyway, I would appreciate a quick solution 'cause my boss
is already on my tail...
Issue History
Date Modified Username Field Change
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07-30-07 13:11 cstadlmann Note Added: 0068070
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