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<seandar...@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@from-outside:1] NoOp("IAX2/iaxtest-1945", "") in new stack
    -- Executing [123@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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