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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users