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

[sean darcy]

On 03/09/2012 04:16 PM, sean darcy wrote:
> 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
>

So I tried having teliax connect to the asterisk box with iax. But now I 
get no audio both ways!

        Answer("IAX2/iaxtest-1945", "") in new stack
        GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack

     -- Goto (from-outside,123,1)
     -- Executing [123 at from-outside:1] NoOp("IAX2/iaxtest-1945", "") in 
new stack
     -- Executing [123 at from-outside:2] Dial("IAX2/iaxtest-1945", 
"SIP/jnctn/1aaabbbcccc") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
     -- Called SIP/jnctn/1aaabbbcccc
     -- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-00000000
     -- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-00000000
     -- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-00000000
     -- SIP/jnctn-00000000 is ringing
     -- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-00000000
     -- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-00000000
     -- SIP/jnctn-00000000 answered IAX2/iaxtest-1945

Really puzzled.

sean