Thanks Brian, but we know arrangement <param name="auth-calls" value="false"/> in internal.xml, and it works in previous releases. But that's not enough now ... or it's a bug ;) Before arrangement ( it is copied from the protocol ) 2008-09-02 13:43:35:691 1220352215.691228: Aborting call on unexpected message for Call-Id '79-5239@127.0.0.1': while expecting '180' (index 2), received 'SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-5239-79-0;received=192.168.2.107 From: sipp <sip:sipp@127.0.0.1:5061>;tag=5239SIPpTag0079 To: sut <sip:2000@192.168.2.107:5060>;tag=3KKN20eKpFFrN Call-ID: 79-5239@127.0.0.1 ... - after arrangement of auth-calls : 2008-09-02 14:38:05:582 1220355485.582747: Aborting call on unexpected message for Call-Id '1-5566@127.0.0.1': while expecting '180' (index 2), received 'SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-5566-1-0;received=192.168.2.107 From: sipp <sip:sipp@127.0.0.1:5061>;tag=5566SIPpTag001 To: sut <sip:2000@192.168.2.107:5060>;tag=2Qv35F85NtN9D Call-ID: 1-5566@127.0.0.1 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9377 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Content-Length: 0 # ./sipp -sn uac -s 2000 192.168.2.107:5070 - all OK # ./sipp -sn uac -s 2000 192.168.2.107:5080 - all OK # ./sipp -sn uac -s 2000 192.168.2.107 - 403 Forbidden ----- Original Message ----- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, September 02, 2008 2:25 PM Subject: Re: [Freeswitch-users] How to configure FreeSWITCH for calling SIPprofile "internal"? auth-calls set to false /b On Sep 2, 2008, at 6:21 AM, Евгений Золотов wrote: How I should reconstruct configuration files of FreeSWITCH, that any user could carry out call at SIP profile "internal" port 5060 and also it was not required to its registration (message REGISTER from one)? Like at profiles "external" and "nat" ( version 1.0.trunk( 9377 ) - earlier these profiles were called in another way: "default" and "outbound"?). ) It's necessary for tests - calls from SIPP client's side, which in standard scenarios does not cause REGISTER and carries out INVITE with a name "sipp". With gratitude, Evgeniy. Brian West sip:brian@freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org