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

[sean darcy]

On 03/09/2012 09:42 PM, Arstan Jusupov wrote:
> 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
>>

The instance firewall is flushed. The security group allows udp 
10000-20000 , 5060 and 4569.

Well it gets stranger:

I set up a sip link to my home. Dialed the teliax number from my cell. 
Asterisk used the sip link to my home - and that worked!

Dial("IAX2/iaxtest-584", "SIP/sip-to-home")

Which seems to mean that the teliax <-> asterisk link is fine.

But if I use a SIP/PSTN provider , I get one-way audio:

Dial("IAX2/iaxtest-515", "SIP/jnctn/<home-pstn>")

Completely baffled.

sean