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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: "asterisk" <sip:asterisk at x.x.x.x>;tag=as7d8aae9d To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> Contact: <sip:asterisk at 203.211.60.167> Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x 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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: "asterisk" <sip:asterisk at 203.211.60.167>;tag=as7d8aae9d To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> Contact: <sip:asterisk at x.x.x.x> Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x 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=888 at mb_tutorial ---------------
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 <sabharwal_rakesh at yahoo.co.uk> 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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:asterisk at x.x.x.x>;tag=as7d8aae9d > To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:asterisk at 203.211.60.167> > Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x > 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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:asterisk at 203.211.60.167>;tag=as7d8aae9d > To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:asterisk at x.x.x.x> > Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x > 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=888 at 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 >
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 <darrin.asterisk at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> 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 <sabharwal_rakesh at yahoo.co.uk> 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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:asterisk at x.x.x.x>;tag=as7d8aae9d > To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:asterisk at 203.211.60.167> > Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x > 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:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport > From: "asterisk" <sip:asterisk at 203.211.60.167>;tag=as7d8aae9d > To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP> > Contact: <sip:asterisk at x.x.x.x> > Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x > 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=888 at 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