I ever met same issue in EC2. <div><br></div><div>Modify the sip_profile configuration file like &quot;internal.xml&quot;</div><div>Replace the below item with actual ip address:</div><div><div>&lt;param name=&quot;ext-rtp-ip&quot; value=&quot;xxx.xxx.xxx.xxx&quot;/&gt;</div>
<div><br></div><div>Thanks,</div><div>Will</div><br><div class="gmail_quote">On Fri, Jul 9, 2010 at 7:31 AM, paul gore <span dir="ltr">&lt;<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I got ngrep trace for port 5060 while making a call to a US number via <a href="http://siptraffic.com" target="_blank">siptraffic.com</a> from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing.<br>
Is there anything in this trace which can help identify the problem?<br>
<br><a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a> - is my local EC2 IP<br><a href="http://184.72.206.204:5060" target="_blank">184.72.206.204:5060</a> - is my public EC2 IP<br><a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a> - <a href="http://siptraffic.com" target="_blank">siptraffic.com</a> proxy IP<br>

<br>Thanks!<br><br><br><br> <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a> -&gt; <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a><br>  INVITE <a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f<br>

  524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: &lt;<a href="http://sip:4000002@67.33.160.119:18027" target="_blank">sip:4000002@67.33.160.119:18027</a>&gt;..To: &quot;45517<br>  709248570&quot;&lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;..From: &quot;4000002&quot;&lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..Call-<br>

  ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE<br>  , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user<br>

  name=&quot;4000002&quot;,realm=&quot;<a href="http://myserver.com" target="_blank">myserver.com</a>&quot;,nonce=&quot;cf9019cc-f44a-4568-97d1-e9883fb1821f&quot;,uri=&quot;sip:45517705678570@v<br>  <a href="http://ersafon.com" target="_blank">ersafon.com</a>&quot;,response=&quot;57da72527524e0e065c9a3221bfadd38&quot;,cnonce=&quot;140f655ff3427f6ba3767ab7040231f3&quot;,nc=0000<br>

  0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=-<br>   8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107<br>

  119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107<br>   BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap<br>

  :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv..<br>#<br>U <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a> -&gt; <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a><br>

  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r<br>  port=18294;received=67.33.160.119..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..To: &quot;455177092<br>

  48570&quot; &lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I<br>  NVITE..User-Agent: myserver..Content-Length: 0....<br>

#<br>U <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a> -&gt; <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a><br>  INVITE <a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9h<br>

  G4bKBU626KBp16t5Q..Max-Forwards: 69..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j.<br>  .To: &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a>&gt;..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647<br>

  2 INVITE..Contact: &lt;sip:gw+<a href="http://voicetrading.com" target="_blank">voicetrading.com</a>@184.72.206.204:5080;transport=udp;gw=<a href="http://voicetrading.com" target="_blank">voicetrading.com</a>&gt;..User-A<br>
  gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY..<br>
  Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp..<br>  Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: &quot;4000002<br>

  &quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039<br>  1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8<br>

  3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80<br>  00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20..<br>#<br>U <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a> -&gt; <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a><br>

  SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From<br>  : &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j..To: &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.co" target="_blank">sip:0017705678570@sip.siptraffic.co</a><br>

  m&gt;;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c<br>  381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C<br>

  ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C<br>  ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672<br>

  RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20..<br>#<br>U <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a> -&gt; <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a><br>

  SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f524431af92cef56-1-<br>  -d87543-;rport=18294;received=67.33.160.119..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..To:<br>

  &quot;45517705678570&quot; &lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk<br>  zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: &lt;sip:45517705678570@184.72.206.204:5060;transport=udp&gt;..User-A<br>

  gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>  REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events:<br>

   talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-<br>  summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P<br>

  arty-ID: &quot;45517705678570&quot; &lt;<a href="mailto:sip%3A45517705678570@10.194.206.102" target="_blank">sip:45517705678570@10.194.206.102</a>&gt;;party=calling;privacy=off;screen=no....v=0..<br>  o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=<br>

  audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil<br>  enceSupp:off - - - -..a=ptime:20..<br>#<br><br><br><br>U <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a> -&gt; <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a><br>

  SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From<br>  : &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j..To: &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.co" target="_blank">sip:0017705678570@sip.siptraffic.co</a><br>

  m&gt;;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c<br>  381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C<br>

  ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C<br>  ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio<br>

  57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20..<br>#<br><br>U <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a> -&gt; <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a><br>

  ....<br>#<br>U <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a> -&gt; <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a><br>  CANCEL <a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f<br>

  524431af92cef56-1--d87543-;rport..To: &quot;45517705678570&quot;&lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;..From: &quot;4000002&quot;&lt;s<br>  <a href="mailto:ip%3A4000002@myserver.com" target="_blank">ip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC<br>

  EL..Proxy-Authorization: Digest username=&quot;4000002&quot;,realm=&quot;<a href="http://myserver.com" target="_blank">myserver.com</a>&quot;,nonce=&quot;cf9019cc-f44a-4568-97d1-e98<br>  83fb1821f&quot;,uri=&quot;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&quot;,response=&quot;46c7e289f7490c807565c561699b03d6&quot;,cnonce=&quot;a226c<br>

  55446c605ee229f045602b29135&quot;,nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41<br>  150..Content-Length: 0....<br>#<br>U <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a> -&gt; <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a><br>

  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport<br>  =18294;received=67.33.160.119..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..To: &quot;4551770567857<br>

  0&quot; &lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY<br>  mY...CSeq: 2 CANCEL..Content-Length: 0....<br>

#<br>U <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a> -&gt; <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a><br>  SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f524431af92cef56-<br>

  1--d87543-;rport=18294;received=67.33.160.119..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..To<br>  : &quot;45517705678570&quot; &lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN<br>

  jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA<br>  GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla<br>

  ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen<br>  ce.winfo, message-summary, refer..Content-Length: 0....<br>#<br>U <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a> -&gt; <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a><br>

  CANCEL <a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9h<br>  G4bKBU626KBp16t5Q..Max-Forwards: 69..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j.<br>

  .To: &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a>&gt;..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647<br>  2 CANCEL..Reason: Q.850;cause=16;text=&quot;NORMAL_CLEARING&quot;..Content-Length: 0....<br>

#<br><br>U <a href="http://67.33.160.119:18294" target="_blank">67.33.160.119:18294</a> -&gt; <a href="http://10.194.206.102:5060" target="_blank">10.194.206.102:5060</a><br>  ACK <a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486;branch=z9hG4bK-d87543-f524<br>

  431af92cef56-1--d87543-;rport..To: &quot;45517705678570&quot; &lt;<a href="mailto:sip%3A45517705678570@myserver.com" target="_blank">sip:45517705678570@myserver.com</a>&gt;;tag=BXgB1FZBUZ3Da..F<br>  rom: &quot;4000002&quot;&lt;<a href="mailto:sip%3A4000002@myserver.com" target="_blank">sip:4000002@myserver.com</a>&gt;;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm<br>

  Y...CSeq: 2 ACK..Content-Length: 0....<br>#<br><br>U <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a> -&gt; <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a><br>  SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: &quot;4000002&quot; &lt;s<br>

  <a href="mailto:ip%3A0014444295793@184.72.206.204" target="_blank">ip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j..To: &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a>&gt;..Contact: s<br>

  ip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL<br>  ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA<br>

  GE..Content-Length: 0....<br>#<br>U <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a> -&gt; <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a><br>  SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..Fr<br>

  om: &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j..To: &lt;sip:0017705678570@sip.siptraffic.<br>  com&gt;..Contact: sip:0017705678570@77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq:<br>

  133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP<br>  TIONS,INFO,MESSAGE..Content-Length: 0....<br>#<br>U <a href="http://10.194.206.102:5080" target="_blank">10.194.206.102:5080</a> -&gt; <a href="http://77.72.169.128:5060" target="_blank">77.72.169.128:5060</a><br>

  ACK <a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a> SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4b<br>  KBU626KBp16t5Q..Max-Forwards: 69..From: &quot;4000002&quot; &lt;<a href="mailto:sip%3A0014444295793@184.72.206.204" target="_blank">sip:0014444295793@184.72.206.204</a>&gt;;tag=18853e82KDe7j..To<br>

  : &lt;<a href="mailto:sip%3A0017705678570@sip.siptraffic.com" target="_blank">sip:0017705678570@sip.siptraffic.com</a>&gt;..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A<br>  CK..Content-Length: 0....<br>
#<br><br><br>
<br><br><br><br><br><br><div class="gmail_quote">On Wed, Jul 7, 2010 at 5:50 PM, paul gore <span dir="ltr">&lt;<a href="mailto:paul.gore.j@gmail.com" target="_blank">paul.gore.j@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">

Seems like siptraffic uses 6 ip addresses for media, can that be the<br>
problem? Is there any setting in a gateway config xml which helps with<br>
that?<br>
I will do ngrep thing and update.<br>
<div><div></div><div><br>
On 7/7/10, paul gore &lt;<a href="mailto:paul.gore.j@gmail.com" target="_blank">paul.gore.j@gmail.com</a>&gt; wrote:<br>
&gt; This provider does work on another box which is not natted as ec2.<br>
&gt; Most puzzling here though is why call originaion via api even not<br>
&gt; going via siptraffic still gets no audio.<br>
&gt;<br>
&gt; On 7/7/10, Tony Graziano &lt;<a href="mailto:tgraziano@myitdepartment.net" target="_blank">tgraziano@myitdepartment.net</a>&gt; wrote:<br>
&gt;&gt; You should try from a standalone or local installation to ensure it works<br>
&gt;&gt; with this provider and your account before you attempt to run it on ec2<br>
&gt;&gt; (imo).<br>
&gt;&gt;<br>
&gt;&gt; On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin<br>
&gt;&gt; &lt;<a href="mailto:sos@sokhapkin.dyndns.org" target="_blank">sos@sokhapkin.dyndns.org</a>&gt;wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; What &quot;doesn&#39;t work&quot; means? It could be (and most likely is not)<br>
&gt;&gt;&gt; FS-related<br>
&gt;&gt;&gt; problem<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Wednesday 07 July 2010, Madovsky wrote:<br>
&gt;&gt;&gt; &gt; I had same problem from this provider without to explain why.<br>
&gt;&gt;&gt; &gt; One day it works, another day it doesn&#39;t, their support is crap...<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;   ----- Original Message -----<br>
&gt;&gt;&gt; &gt;   From: Anthony Minessale<br>
&gt;&gt;&gt; &gt;   To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
&gt;&gt;&gt; &gt;   Sent: Wednesday, July 07, 2010 2:37 PM<br>
&gt;&gt;&gt; &gt;   Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on<br>
&gt;&gt;&gt; &gt; outgoing<br>
&gt;&gt;&gt; &gt;  calls<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;   not really, not with so little information.<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;   On Wed, Jul 7, 2010 at 1:30 PM, paul gore &lt;<a href="mailto:paul.gore.j@gmail.com" target="_blank">paul.gore.j@gmail.com</a>&gt;<br>
&gt;&gt;&gt; wrote:<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;     Firewall is configured according to the wiki, I also tried to open<br>
&gt;&gt;&gt; all<br>
&gt;&gt;&gt; &gt;     udp ports, issue persists.<br>
&gt;&gt;&gt; &gt;     Actually the problem became more complex - outgoing calls don&#39;t<br>
&gt;&gt;&gt; &gt; work<br>
&gt;&gt;&gt; &gt;     with one particular termination provider, <a href="http://siptraffic.com" target="_blank">siptraffic.com</a> , any<br>
&gt;&gt;&gt; &gt; ideas<br>
&gt;&gt;&gt; &gt;     why?<br>
&gt;&gt;&gt; &gt;     Outgoing calls also don&#39;t work when originating a call via js<br>
&gt;&gt;&gt; &gt; script<br>
&gt;&gt;&gt; &gt;     or via FS api. Any clues on that one?<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;     On 7/6/10, paul gore &lt;<a href="mailto:paul.gore.j@gmail.com" target="_blank">paul.gore.j@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt; &gt;     &gt; Hi there,<br>
&gt;&gt;&gt; &gt;     &gt; I am experimenting with FS on EC2, I like results, but stuck on<br>
&gt;&gt;&gt; weird<br>
&gt;&gt;&gt; &gt;     &gt; audio issue - I followed FreeSwitch EC2 wiki article and<br>
&gt;&gt;&gt; &gt; modified<br>
&gt;&gt;&gt; &gt;     &gt; internal profile<br>
&gt;&gt;&gt; &gt;     &gt; and vars.xml accordingly, but unfortunately still cannot get it<br>
&gt;&gt;&gt; &gt;     &gt; working. Incoming and outgoing calls made using a SIP phone to<br>
&gt;&gt;&gt; &gt; FS<br>
&gt;&gt;&gt; &gt;     &gt; extensions work just fine. As well as calls to FS from PSTN. But<br>
&gt;&gt;&gt; &gt;     &gt; calls to PSTN via gateways result in no audio at all, no ring,<br>
&gt;&gt;&gt; &gt;     &gt; nothing, SIP signaling goes through OK. Sofia status profile<br>
&gt;&gt;&gt; &gt; shows<br>
&gt;&gt;&gt; &gt;     &gt; correct values for Ext-RTP-IP for both profiles -<br>
&gt;&gt;&gt; &gt;     &gt; my static public IP, RTP-IP shows local IP.<br>
&gt;&gt;&gt; &gt;     &gt; Any thoughts on that? Anybody can share working profile<br>
&gt;&gt;&gt; configuration<br>
&gt;&gt;&gt; &gt;     &gt; may be?<br>
&gt;&gt;&gt; &gt;     &gt; Please help, I really need to get this going.<br>
&gt;&gt;&gt; &gt;     &gt;<br>
&gt;&gt;&gt; &gt;     &gt; Thanks.<div class="im"><br>
&gt;&gt;&gt; &gt;<br>
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&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a>&lt;<a href="mailto:PAYPAL%253Aanthony.minessale@gmail.com" target="_blank">PAYPAL%3Aanthony.minessale@gmail.com</a>&gt;<br>


&gt;&gt;&gt; &gt;   IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;   FreeSWITCH Developer Conference<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a>&lt;<a href="mailto:sip%253A888@conference.freeswitch.org" target="_blank">sip%3A888@conference.freeswitch.org</a>&gt;<br>

&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt; <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a>&lt;<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org" target="_blank">googletalk%3Aconf%2B888@conference.freeswitch.org</a>&gt;<br>


&gt;&gt;&gt; &gt;   pstn:+19193869900<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; ---------------------------------------------------------------------------<br>
&gt;&gt;&gt; &gt; ---<div class="im"><br>
&gt;&gt;&gt; &gt;<br>
&gt;&gt;&gt; &gt;<br>
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&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br></div>
&gt;&gt; --<br>
&gt;&gt; ======================<br>
&gt;&gt; Tony Graziano, Manager<br>
&gt;&gt; Telephone: 434.984.8430<br>
&gt;&gt; sip: <a href="mailto:tgraziano@voice.myitdepartment.net" target="_blank">tgraziano@voice.myitdepartment.net</a><br>
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&gt;&gt;<br>
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&gt;&gt;<br>
&gt;&gt; LAN/Telephony/Security and Control Systems Helpdesk:<br>
&gt;&gt; Telephone: 434.984.8426<br>
&gt;&gt; sip: <a href="mailto:helpdesk@voice.myitdepartment.net" target="_blank">helpdesk@voice.myitdepartment.net</a><br>
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&gt;&gt;<br>
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&gt;&gt; Because 31 Oct = 25 Dec.<br>
&gt;&gt;<br>
&gt;<br>
</div></div></blockquote></div><br>
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<br></blockquote></div><br></div>