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

sean darcy seandarcy2 at gmail.com
Fri Mar 9 15:16:01 CST 2012

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:

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)

exten => 123,1,NoOp()
exten => 123,n,Answer()
exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy)
exten => 123,n,HangUp()

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/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 

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

Any help really appreciated!


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