[asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

I'm trying to move the asterisk server to an Amazon Web instance. We 
have teliax for our sip provider. I'd like for our DID lines to be 
connected to a users cell phone.

Seems simple enough, but I'm getting the dreaded one-way audio, even 
with nat=yes everyplace I can think of.

The dialplan is real easy:

[from-teliax-sip]
exten => _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten => _j.,n,Set(3digitexten=${EXTEN:12:3}
exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten => _j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten => 123,1,NoOp()
exten => 123,n,Answer()
exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy)
exten => 123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no  ; tried nonat

sip show peer jnctn:
   Insecure     : invite
   Force rport  : Yes
   .........
   DirectMedia  : No

sip show peer teliax:
   Insecure     : port,invite
   Force rport  : Yes
   ........
   DirectMedia  : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-00000022", ""From teliax sip with exten 
"<somename12lg>(123)"") in new stack
Set("SIP/teliax-00000022", "3digitexten=123") in new stack
NoOp("SIP/teliax-00000022", ""Callerid is " "") in new stack
Goto("SIP/teliax-00000022", "from-outside,123,1") in new stack
     -- Goto (from-outside,123,1)
NoOp("SIP/teliax-00000022", "") in new stack
Answer("SIP/teliax-00000022", "") in new stack
Dial("SIP/teliax-00000022", "SIP/jnctn/1212aaabbbb") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
     -- Called SIP/jnctn/1212aaabbbb
     -- SIP/jnctn-00000023 is making progress passing it to 
SIP/teliax-00000022
     -- SIP/jnctn-00000023 answered SIP/teliax-00000022
     -- Locally bridging SIP/teliax-00000022 and SIP/jnctn-00000023
   == Spawn extension (from-outside, 123, 3) exited non-zero on 
'SIP/teliax-00000022'

The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean