If you want anonymous callers to be able to place calls to Asterisk, you need to set allowguest=yes.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis Sent: Saturday, April 23, 2011 9:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cannot call to my server with SIP Op 22-04-11 23:49, Jamie A. Stapleton schreef: > I can see your server just fine... > > -bash-3.2# ./svmap.py xen8.vandervlis.nl > | SIP Device | User Agent | Fingerprint | > ---------------------------------------------------------------------- > | 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled | > > However, if I try to call, Asterisk is saying: > -- Called p...@vandervlis.nl > [2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: > Failed to authenticate on INVITE to ...;tag=as131f7b6a' Ah, this is very good information. I see you, but I don't understand why I don't see myself when I try this. Maybe my sip client (Ekiga) is not OK. Asterisk log: [Apr 22 23:46:50] NOTICE[29497] chan_sip.c: Sending fake auth rejection for device "Jamie A. Stapleton" <sip:2233440...@sip2sip.info>;tag=0wqaLzsAyMQwTdfcP2r0mG2FkPBQjEQF Firewall log: Apr 22 23:46:50 xen8 kernel: [3824476.043190] FW:IN=eth0 OUT= MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150 DST=91.198.178.28 LEN=1320 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP SPT=5060 DPT=5060 LEN=1300 Apr 22 23:46:50 xen8 kernel: [3824476.043556] FW:IN= OUT=eth0 SRC=91.198.178.28 DST=81.23.228.150 LEN=782 TOS=0x00 PREC=0x00 TTL=64 ID=17809 PROTO=UDP SPT=5060 DPT=5060 LEN=762 Apr 22 23:46:50 xen8 kernel: [3824476.048153] FW:IN=eth0 OUT= MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150 DST=91.198.178.28 LEN=411 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP SPT=5060 DPT=5060 LEN=391 > What do you have allowguest > (http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to? I was testing security. It's like this: sip.conf: ----------- [general] context=default allowguest=no alwaysauthreject=yes (...) [guests] context=default allowguest=yes [trunk] context=dialout (...) [phone-paul] context=dialout (...) [phone-ann] context=dialout (...) ----------- extensions.conf: ------------- [default] include => users [dialout] include => users exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT) [users] exten=>6001,1,Dial(SIP/paul,20) exten=>6002,1,Dial(SIP/ann,20) (...) -------- Thanks for your help! With regards, Paul van der Vlis. -- http://www.vandervlis.nl/ -- _____________________________________________________________________ -- 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