<p>Have you looked at rtp debug? Is it possible reinvites are enabled?</p>
<div class="gmail_quote">On Mar 9, 2012 9:20 PM, "sean darcy" <<a href="mailto:seandarcy2@gmail.com">seandarcy2@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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?<br>
<br>
Sent from my iPhone<br>
<br>
On Mar 10, 2012, at 7:16 AM, sean darcy<<a href="mailto:seandarcy2@gmail.com" target="_blank">seandarcy2@gmail.com</a>> wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On 03/09/2012 04:16 PM, sean darcy wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I'm trying to move the asterisk server to an Amazon Web instance. We<br>
have teliax for our sip provider. I'd like for our DID lines to be<br>
connected to a users cell phone.<br>
<br>
Seems simple enough, but I'm getting the dreaded one-way audio, even<br>
with nat=yes everyplace I can think of.<br>
<br>
The dialplan is real easy:<br>
<br>
[from-teliax-sip]<br>
exten => _j.,1,NoOp("From teliax sip with exten "${EXTEN}")<br>
exten => _j.,n,Set(3digitexten=${EXTEN:<u></u>12:3}<br>
exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} )<br>
exten => _j.,n,GoTo(from-outside,${<u></u>3digitexten},1)<br>
<br>
[from-outside]<br>
exten => 123,1,NoOp()<br>
exten => 123,n,Answer()<br>
exten => 123,n,Dial(SIP/jnctn/<u></u>1212xxxyyyy)<br>
exten => 123,n,HangUp()<br>
<br>
sip.conf:<br>
[general]<br>
externaddr=xx.yyy.zz.aa<br>
nat=yes<br>
directmedia=no ; tried nonat<br>
<br>
sip show peer jnctn:<br>
Insecure : invite<br>
Force rport : Yes<br>
.........<br>
DirectMedia : No<br>
<br>
sip show peer teliax:<br>
Insecure : port,invite<br>
Force rport : Yes<br>
........<br>
DirectMedia : No<br>
<br>
<br>
<br>
And the cli doesn't show any problems:<br>
<br>
NoOp("SIP/teliax-00000022", ""From teliax sip with exten<br>
"<somename12lg>(123)"") in new stack<br>
Set("SIP/teliax-00000022", "3digitexten=123") in new stack<br>
NoOp("SIP/teliax-00000022", ""Callerid is " "") in new stack<br>
Goto("SIP/teliax-00000022", "from-outside,123,1") in new stack<br>
-- Goto (from-outside,123,1)<br>
NoOp("SIP/teliax-00000022", "") in new stack<br>
Answer("SIP/teliax-00000022", "") in new stack<br>
Dial("SIP/teliax-00000022", "SIP/jnctn/1212aaabbbb") in new stack<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/jnctn/1212aaabbbb<br>
-- SIP/jnctn-00000023 is making progress passing it to SIP/teliax-00000022<br>
-- SIP/jnctn-00000023 answered SIP/teliax-00000022<br>
-- Locally bridging SIP/teliax-00000022 and SIP/jnctn-00000023<br>
== Spawn extension (from-outside, 123, 3) exited non-zero on<br>
'SIP/teliax-00000022'<br>
<br>
The called party can hear the calling party, but not the reverse!<br>
<br>
Any help really appreciated!<br>
<br>
sean<br>
<br>
</blockquote>
<br>
So I tried having teliax connect to the asterisk box with iax. But now I get no audio both ways!<br>
<br>
Answer("IAX2/iaxtest-1945", "") in new stack<br>
GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack<br>
<br>
-- Goto (from-outside,123,1)<br>
-- Executing [123@from-outside:1] NoOp("IAX2/iaxtest-1945", "") in new stack<br>
-- Executing [123@from-outside:2] Dial("IAX2/iaxtest-1945", "SIP/jnctn/1aaabbbcccc") in new stack<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/jnctn/1aaabbbcccc<br>
-- IAX2/iaxtest-1945 requested special control 20, passing it to SIP/jnctn-00000000<br>
-- IAX2/iaxtest-1945 requested special control 20, passing it to SIP/jnctn-00000000<br>
-- IAX2/iaxtest-1945 requested special control 20, passing it to SIP/jnctn-00000000<br>
-- SIP/jnctn-00000000 is ringing<br>
-- IAX2/iaxtest-1945 requested special control 20, passing it to SIP/jnctn-00000000<br>
-- IAX2/iaxtest-1945 requested special control 20, passing it to SIP/jnctn-00000000<br>
-- SIP/jnctn-00000000 answered IAX2/iaxtest-1945<br>
<br>
Really puzzled.<br>
<br>
sean<br>
</blockquote></blockquote>
<br>
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.<br>
<br>
Flushed the instance iptables, which fixed a problem I was having with a phone registering.<br>
<br>
But I still have my one-way audio. The calling party hears nothing from the called party.<br>
<br>
sean<br>
<br>
<br>
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</blockquote></div>