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<body class='hmmessage'><div dir='ltr'>Hi Matthew,<div><br></div><div>Indeed I missed your previous message!</div><div>After changing the externip, it worked successfully... The sip session is established with the complete <span style="font-size: 12pt;"> </span><span style="font-size: 12pt;">three-way handshake, and the voice packet is exchanged with no problem!</span></div><div><span style="font-size: 12pt;"><br></span></div><div><span style="font-size: 12pt;">Many thanks. </span><span style="font-size: 12pt;"> </span></div><div><br><div>> Date: Fri, 20 Sep 2013 10:01:52 -0500<br>> From: mroth@imminc.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] The call is established but without exchanged voice packets<br>> <br>> Asmaa, <br>> <br>> You're getting ahead of yourself. How do you expect audio to work if<br>> your firewall/NAT settings aren't even configured correctly to<br>> establish SIP sessions?<br>> <br>> Go back and read the message that I sent yesterday. Fix the SIP <br>> three-way handshake problem. That is step 1 and you'll know you have<br>> it right when you stop seeing 'Retransmission timeout reached on<br>> transmission' errors.<br>> <br>> You still won't have audio but that's step 2. It requires properly<br>> configuring Asterisk's NAT settings and the firewall(s) between the<br>> phones and the server to allow RTP traffic to flow, but don't worry<br>> about it until step 1 is complete.<br>> <br>> Regards,<br>> <br>> Matthew Roth<br>> InterMedia Marketing Solutions<br>> Software Engineer and Systems Developer<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div></div> </div></body>
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