Summary:ASTERISK-16791: 1.4 Passthru Receives T38 Re-Invite and passes on a non-T38 invite to other peer.
Reporter:alexr1 (alexr1)Labels:
Date Opened:2010-10-12 01:05:10Date Closed:2011-07-26 15:30:03
Versions:Frequency of
Environment:Attachments:( 0) fail.txt
( 1) failwithpatch.txt
( 2) working.txt
Description:INCOMING Fax from PSTN. ENSR has full T38 Gateway support.

Analogue fax -> PSTN -> ENSR2.5.4 -> Ast1.4.37 -> Ast1.6.1.20

The problem happens when 1.6 re-invites with a T38 Offer to 1.4.
1.4 see's the T38 offer (and sends TRYING to 1.6), but re-invites to ENSR with G711 instead of T38.

Upon close inspection of the Debug log, Asterisk sends TRYING and then chooses to "DEFER" sending the T38 re-invite to ENSR. What happens after that, I'm not exactly sure.


Asterisk 1.4.37 has realtime enabled. The Asterisk peer (on the 1.4 box) is REALTIME, NAT=YES, CANREINVITE=NO. Very similar result if I invert these values.

Attached is two logs. "Working.txt" and "Fail.txt". Fail.txt is a logfile of what happens in the above described scenario. "Working.txt" is what happens when I sent a fax to a non-realtime, non-nat, canreinvite=yes peer (Running Asterisk with ReceiveFax app)
Comments:By: alexr1 (alexr1) 2010-10-13 18:43:48

Bounty: $250 USD via Paypal to the person who can solve this problem.

By: Ramon Peek-Fares (ramonpeek) 2010-10-14 02:33:55

Try patching the 1.4 system with the patch I created in ASTERISK-16658 ;-)

By: alexr1 (alexr1) 2010-10-14 20:09:12

We're getting somewhere. Previously all the caller heard was silence, but now after a couple of seconds of silence, fax tones come on (Which I thought meant a successful T.38 handshake)

Analogue fax -> PSTN -> ENSR2.5.4 -> Ast1.4.37 -> Ast1.6.1.20

This time the fax is actually communicating via T38 but appears to fail after some time.

By: alexr1 (alexr1) 2010-10-14 20:09:40

Bounty increased to $300 USD via Paypal

By: Ramon Peek-Fares (ramonpeek) 2010-10-15 01:12:35

Since I don't have your setup (and can't replicate it), could you send me the log file of Asterisk (debug & sip debugging enabled) + a pCAP trace of the network traffic?
I really need it in order to figure out whats happening..

By: Matthew Nicholson (mnicholson) 2011-07-26 15:29:55.020-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If this is still a problem in 1.8, open a new issue.