Hi Darrin Thanks for your kind reply.
Your description is right, PC(Soft Phone) > ADSL Router > Internet > Asterisk box Thanks for your suggestion on the security. Please advise , I am specifically concerned about the port to which server reply after initial communication (random above 32000) Retransmitting #3 (NAT) to x.x.x.x:38155: Thanks in advance Rakesh ----- Original Message ---- From: Darrin Henshaw <[email protected]> To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Sent: Friday, 16 October, 2009 21:31:47 Subject: Re: [asterisk-users] Soft phone not registering First suggestion is if this Asterisk server is accessible from the internet put a secret in the peer definition. What you have now is wide open. Second thing is if I understand it you are going: PC(Soft Phone) > ADSL Router > Internet > Asterisk box. Is that correct? If not, can you descibe it better. On Fri, Oct 16, 2009 at 7:56 AM, Rakesh Sabharwal <[email protected]> wrote: > > HI All, > > I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying > to connect from softphone behind ADSL router. > > The softphone is not able to register, we get some SIP messages on the > server, which look like below. > > Please advise where could be the issue. > > Thnx > Rakesh > > --- > Retransmitting #3 (NAT) to x.x.x.x:38155: > OPTIONS sip:[email protected]:5060;rinstance=5b19b87f10954011;transport=UDP > SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:[email protected]>;tag=as7d8aae9d > To: <sip:[email protected]:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 16 Oct 2009 10:47:56 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces > Content-Length: 0 > > > --- > Retransmitting #4 (NAT) to x.x.x.x:38155: > OPTIONS sip:[email protected]:5060;rinstance=5b19b87f10954011;transport=UDP > SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:[email protected]>;tag=as7d8aae9d > To: <sip:[email protected]:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 16 Oct 2009 10:47:56 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces > Content-Length: 0 > > -------------------- > > sip.conf ---- > > [general] > context = tutorial > bindport = 5060 > bindaddr =0.0.0.0 > domain = x.x.x.x > nat=yes > disallow = all > allow = alaw > keeprtpalive = yes > notifyringing = yes > canreinvite = no > type = peer > realm = asterisk > qualify = yes > > [test2] > type = peer > host = dynamic > username = test2 > context = tutorial > port = 5060 > notifyringing = yes > nat = yes > type = friend > canreinvite = no > realm = asterisk > qualify = yes > mailbox=...@mb_tutorial > > --------------- > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
