Have you looked at rtp debug? Is it possible reinvites are enabled? On Mar 9, 2012 9:20 PM, "sean darcy" <seandar...@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<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 > > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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