It does not appear to make any difference.  Calls are still failing.

-----Original Message-----
From: Christopher Harrington [mailto:[email protected]] 
Sent: Thursday, December 27, 2012 1:20 PM
To: Eric Wieling
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 
Pass-through

True, but it should bypass Asterisk when possible for SIP streams and may solve 
your problem.


On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling <[email protected]> wrote:


        We have directrtpsetup=no because the comments in the sample config 
indicates it does not work in all situations.
        

        -----Original Message-----
        From: [email protected] 
[mailto:[email protected]] On Behalf Of Christopher 
Harrington
        Sent: Thursday, December 27, 2012 1:13 PM
        To: Asterisk Users Mailing List - Non-Commercial Discussion
        Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 
Pass-through
        
        directrtpsetup=yes in sip.conf?
        
        
        
        On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling <[email protected]> 
wrote:
        
        
                We have set directmedia=yes as well as directmedia=no.  There 
is no NAT involved.
        
        
        
        
                -----Original Message-----
                From: [email protected] 
[mailto:[email protected]] On Behalf Of Leandro Dardini
                Sent: Thursday, December 27, 2012 1:08 PM
                To: Asterisk Users Mailing List - Non-Commercial Discussion
                Subject: Re: [asterisk-users] $100 Bounty: 
Level3/Asterisk/Adtran T.38 Pass-through
        
                Have you configured the canreinvite=yes in sip peer?
        
                I am currently off work for two days, but a 100% fail means a 
configuration problem for sure.
        
        
                Leandro
        
        
                2012/12/27 Eric Wieling <[email protected]>
        
        
                        We are offering $100 (paid via paypal or check) to the 
first person who assists us in successfully sending and receiving faxes in the 
setup described below.  Offer expires Dec 31.  We are a direct customer of 
Level 3, there is no other carrier involved.
        
                        What we want to work:
        
                            Level 3 T.38 TN <-> MSX/Nextone SBC <-> Asterisk 
1.8.18.1 <-> Adtran NetVanta w/POTS and T.38 support.
        
                        When we replace Asterisk with Kamailio faxes work fine. 
 When we put Asterisk there instead, then faxes fail nearly 100% of the time.
        
                        I see the switch to T.38 in the Adtran debug logs.   We 
can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax 
using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct.
        
        
        
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-Chris Harrington

ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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