[Asterisk-Users] One-Way SIP Audio with SVN Codebase

[George Pajari]

Please tell me the obvious mistake I'm making here. (And yes, I well 
know about NAT and one-way audio problems in general.)

I want to try the new T.38 passthrough stuff, downloaded it, built it, 
tested it with an SPA-2100 and can hear announcements fine but echo test 
shows no audio outbound (i.e. SPA to Asterisk).

Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in 
the same rack (IP address differs by 1) and it works fine (which 
suggests no problem with the NAT router the SPA-2100 is behind).

Here are the various SVN checkouts I did (all have the problem -- if you 
can't guess I'm no SVN guru -- SCCS was the last version control system 
I mastered).

  svn checkout  
http://svn.digium.com/svn/asterisk/team/oej/t38passthrough asterisk
  svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
  svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2

Here is the sip.conf entry:

[t38.nvc.a]
type=friend
context=t38-inbound
callerid=John Doe <1234>
host=dynamic
secret=xxyyzz
qualify=2000
nat=yes
accountcode=NVC06
type=friend
dtmfmode=rfc2833
canreinvite=no

What else do you need? Where do I look? I'm assuming the problem is 
either in the manner I'm checking out the code or in some 1.2-specific 
config thing I'm overlooking (the missing no-one-way-audio option in 
sip.conf?)

-- 
George Pajari, netVOICE communications    604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
                  www.netvoice.ca  www.ip-centrex.ca
      www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca