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

[Arstan Jusupov]

Udp port 5060, udp port range 10000-20000 open? Those are for sip.

For iax2 udp port 4569

Make sure they are open. 

Also can you register two ext from the same instance and see if you can hear both ways....

What kind of trunk do you have to the other side you calling?

Arstan
Sent from my iPhone

On Mar 10, 2012, at 10:20 AM, 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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