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

[Positively Optimistic]

Have you looked at rtp debug?   Is it possible reinvites are enabled?
On Mar 9, 2012 9:20 PM, "sean darcy" <seandarcy2 at gmail.com> wrote:

> On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
>
>> 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
>>>
>>
> Well that's interesting. I hadn't realized that iptables was set up on the
> instance, as well as the firewall from the security group on the control
> panel.
>
> Flushed the instance iptables, which fixed a problem I was having with a
> phone registering.
>
> But I still have my one-way audio. The calling party hears nothing from
> the called party.
>
> sean
>
>
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