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

[sean darcy]

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