So "don't work" still means "no audio" and "sip is ok", clear...<div><br></div><div>RTP packets are lost somewhere on the way:</div><div><br></div><div>You can check where RTP packets are lost - between FS and siptraffic, or between FS and your SIP Client.</div>
<div><br></div><div>1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install ngrep)</div><div>2.Run it: "ngrep port 5080"</div><div>3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C</div>
<div>4.Find the first INVITE line from your FS to siptraffic, it will contain your IP and port at the following lines:</div><div>c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio ****!!!YOUR-RTP-PORT!!!**** RTP/AVP</div>
<div>5.Open new terminal window, and run "ngrep port ****!!!YOUR-RTP-PORT!!!****"</div><div>6. Wait 5 seconds</div><div>7. Stop ngrep using Ctrl+C</div><div>8. Hangup</div><div><br></div><div><div>Now on the second terminal you should see a lot of line pairs like:</div>
<div>YOUR-EC-IP -> SIPTRAFFIC-IP</div><div>SIPTRAFFIC-IP -> YOUR-EC-IP</div></div><div><br></div><div>If you see only one of the directions (e.g. only YOUR-EC-IP -> SIPTRAFFIC-IP), then some problem is between FS and Siptraffic.</div>
<div><br></div><div>If you see both directions then problem is not here and most probably on the way from FS to your SIP Client or somewhere else (inside FS?)</div><div>If so, try to investigate this part using port 5060 (same way as 5080).</div>
<div><br></div><div>This analysis will narrow the problem a bit...</div><div><br></div><div>Regads,</div><div>Vitalie</div><div><br></div><div><br><br><div class="gmail_quote">2010/7/7 paul gore <span dir="ltr"><<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">This provider does work on another box which is not natted as ec2.<br>
Most puzzling here though is why call originaion via api even not<br>
going via siptraffic still gets no audio.<br>
<div><div></div><div class="h5"><br>
On 7/7/10, Tony Graziano <<a href="mailto:tgraziano@myitdepartment.net">tgraziano@myitdepartment.net</a>> wrote:<br>
> You should try from a standalone or local installation to ensure it works<br>
> with this provider and your account before you attempt to run it on ec2<br>
> (imo).<br>
><br>
> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin<br>
> <<a href="mailto:sos@sokhapkin.dyndns.org">sos@sokhapkin.dyndns.org</a>>wrote:<br>
><br>
>> What "doesn't work" means? It could be (and most likely is not) FS-related<br>
>> problem<br>
>><br>
>> On Wednesday 07 July 2010, Madovsky wrote:<br>
>> > I had same problem from this provider without to explain why.<br>
>> > One day it works, another day it doesn't, their support is crap...<br>
>> ><br>
>> > ----- Original Message -----<br>
>> > From: Anthony Minessale<br>
>> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> > Sent: Wednesday, July 07, 2010 2:37 PM<br>
>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing<br>
>> > calls<br>
>> ><br>
>> ><br>
>> > not really, not with so little information.<br>
>> ><br>
>> ><br>
>> ><br>
>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore <<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>><br>
>> wrote:<br>
>> ><br>
>> > Firewall is configured according to the wiki, I also tried to open<br>
>> all<br>
>> > udp ports, issue persists.<br>
>> > Actually the problem became more complex - outgoing calls don't work<br>
>> > with one particular termination provider, <a href="http://siptraffic.com" target="_blank">siptraffic.com</a> , any ideas<br>
>> > why?<br>
>> > Outgoing calls also don't work when originating a call via js script<br>
>> > or via FS api. Any clues on that one?<br>
>> ><br>
>> > On 7/6/10, paul gore <<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>> wrote:<br>
>> > > Hi there,<br>
>> > > I am experimenting with FS on EC2, I like results, but stuck on<br>
>> weird<br>
>> > > audio issue - I followed FreeSwitch EC2 wiki article and modified<br>
>> > > internal profile<br>
>> > > and vars.xml accordingly, but unfortunately still cannot get it<br>
>> > > working. Incoming and outgoing calls made using a SIP phone to FS<br>
>> > > extensions work just fine. As well as calls to FS from PSTN. But<br>
>> > > calls to PSTN via gateways result in no audio at all, no ring,<br>
>> > > nothing, SIP signaling goes through OK. Sofia status profile shows<br>
>> > > correct values for Ext-RTP-IP for both profiles -<br>
>> > > my static public IP, RTP-IP shows local IP.<br>
>> > > Any thoughts on that? Anybody can share working profile<br>
>> configuration<br>
>> > > may be?<br>
>> > > Please help, I really need to get this going.<br>
>> > ><br>
>> > > Thanks.<br>
>> ><br>
>> > _______________________________________________<br>
>> > FreeSWITCH-users mailing list<br>
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>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> ><br>
>> ><br>
>> ><br>
>> ><br>
>> ><br>
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</div></div>>> > <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><<a href="mailto:MSN%253Aanthony_minessale@hotmail.com">MSN%3Aanthony_minessale@hotmail.com</a>><br>
>> ><br>
>> > GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><<a href="mailto:PAYPAL%253Aanthony.minessale@gmail.com">PAYPAL%3Aanthony.minessale@gmail.com</a>><br>
<div class="im">>> > IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
>> ><br>
>> > FreeSWITCH Developer Conference<br>
</div>>> > <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><<a href="mailto:sip%253A888@conference.freeswitch.org">sip%3A888@conference.freeswitch.org</a>><br>
>> ><br>
>> > <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org">googletalk%3Aconf%2B888@conference.freeswitch.org</a>><br>
<div><div></div><div class="h5">>> > pstn:+19193869900<br>
>> ><br>
>> ><br>
>> ><br>
>> ><br>
>> ---------------------------------------------------------------------------<br>
>> > ---<br>
>> ><br>
>> ><br>
>> > _______________________________________________<br>
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>> ><br>
>><br>
>><br>
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>><br>
><br>
><br>
><br>
> --<br>
> ======================<br>
> Tony Graziano, Manager<br>
> Telephone: 434.984.8430<br>
> sip: <a href="mailto:tgraziano@voice.myitdepartment.net">tgraziano@voice.myitdepartment.net</a><br>
> Fax: 434.984.8431<br>
><br>
> Email: <a href="mailto:tgraziano@myitdepartment.net">tgraziano@myitdepartment.net</a><br>
><br>
> LAN/Telephony/Security and Control Systems Helpdesk:<br>
> Telephone: 434.984.8426<br>
> sip: <a href="mailto:helpdesk@voice.myitdepartment.net">helpdesk@voice.myitdepartment.net</a><br>
> Fax: 434.984.8427<br>
><br>
> Helpdesk Contract Customers:<br>
> <a href="http://www.myitdepartment.net/gethelp/" target="_blank">http://www.myitdepartment.net/gethelp/</a><br>
><br>
> Why do mathematicians always confuse Halloween and Christmas?<br>
> Because 31 Oct = 25 Dec.<br>
><br>
<br>
</div></div><div><div></div><div class="h5">_______________________________________________<br>
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</div></div></blockquote></div><br></div>