Hi Matthew, Indeed I missed your previous message!After changing the externip, it worked successfully... The sip session is established with the complete three-way handshake, and the voice packet is exchanged with no problem! Many thanks. > Date: Fri, 20 Sep 2013 10:01:52 -0500 > From: mr...@imminc.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] The call is established but without exchanged > voice packets > > Asmaa, > > You're getting ahead of yourself. How do you expect audio to work if > your firewall/NAT settings aren't even configured correctly to > establish SIP sessions? > > Go back and read the message that I sent yesterday. Fix the SIP > three-way handshake problem. That is step 1 and you'll know you have > it right when you stop seeing 'Retransmission timeout reached on > transmission' errors. > > You still won't have audio but that's step 2. It requires properly > configuring Asterisk's NAT settings and the firewall(s) between the > phones and the server to allow RTP traffic to flow, but don't worry > about it until step 1 is complete. > > Regards, > > Matthew Roth > InterMedia Marketing Solutions > Software Engineer and Systems Developer > > -- > _____________________________________________________________________ > -- 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
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