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

[Arstan Jusupov]

It may sound silly but did you configure/open firewall ports on amazon ec2? The instance itself as we as from the amazon ec2 panel?

Sent from my iPhone

On Mar 10, 2012, at 7:16 AM, sean darcy <seandarcy2 at gmail.com> wrote:

> 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
> 
> 
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