[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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
Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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
Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com mailto:gopalakrishnan...@gmail.com wrote: Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
Thanks for the comments. Without changing anything, adding fromuser=usman02 in both side worked for me.. Thanks. On Mon, Aug 19, 2013 at 1:01 AM, Andrew Colin and...@vsave.co.za wrote: change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
just remove username. type peer authenticate by ip On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin and...@vsave.co.za wrote: change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
ok thanks Asghar Mohammad On Mon, Aug 19, 2013 at 1:05 AM, Asghar Mohammad asghar...@gmail.comwrote: just remove username. type peer authenticate by ip On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin and...@vsave.co.za wrote: change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no use... On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten = _1X.,1,Dial(SIP/usman02/${EXTEN}) exten = _1X.,n,Hangup Server2 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.10.10.81 context=us02-trunk-inbound port=5060 qualify=yes disallow=all allow=g729 ;allow=ulaw ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=port,invite extensions.conf [us02-trunk-inbound] exten = _X.,Dial(SIP/${EXTEN},60) Now when I dial from server1, in the server 2 am getting the error as, [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth: username mismatch, have 2001, digest has usman02 things are fine.. but I dont know where the mistake is...! Can you some one advise me... ! Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
[asterisk-users] Am I being hacked?
Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-00a9] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure out where this attempt is coming from so I can block it. -- Ira-- _ -- 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
Re: [asterisk-users] Am I being hacked?
Hi, for example http://www.fail2ban.org/wiki/index.php/Asterisk On 18 August 2013 23:41, Ira i...@extrasensory.com wrote: Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx ;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-00a9] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx ;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure out where this attempt is coming from so I can block it. -- Ira -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Am I being hacked?
Hi You should install something like fail2ban Regards On Sun, Aug 18, 2013 at 5:41 PM, Ira i...@extrasensory.com wrote: Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx ;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-00a9] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx ;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure out where this attempt is coming from so I can block it. -- Ira -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Am I being hacked?
On Sun, 18 Aug 2013, Ira wrote: [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=2762c06e I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure out where this attempt is coming from so I can block it. Any chance '390' is a legitimate (but mis-configured or obsolete) device on your network? Is xx.xx.xxx.xxx a private or public address? Can you 'wireshark' some packets and see if the OUI matches one of your endpoints? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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