We had a question:
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"?). )
Brian West has advised to us:
auth-calls set to false
But we know arrangement <param name="auth-calls" value="false"/> in internal.xml, and it works in previous releases. In the previous releases: freeswitch@opos8.altron.lan> version FreeSwitch Version 1.0.rc4 (exported) arrangement <param name="auth-calls" value="false"/> in SIP profile default.xml led to successful performance of the command: # ./sipp -sn uac -s 2000 192.168.2.107 (extension 2000 - this is our own extension called application info(), that we also see on console FreeSWITCH). In the last release: freeswitch@opos9.altron.lan> version FreeSwitch Version 1.0.trunk (9377) - now the profile is defined in internal.xml, and default - alias - is does not adduce to same result, and INVITE comes to the end with an error UDP message received [644] bytes : SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-4457-1-0;received=192.168.2.107 From: sipp <sip:sipp@127.0.0.1:5061>;tag=4457SIPpTag001 To: sut <sip:2000@192.168.2.107:5060>;tag=ayUKNQ761p51r Call-ID: 1-4457@127.0.0.1 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9377 And we can see such message on the FreeSWITCH console? 2008-09-03 11:35:57 [WARNING] sofia.c:3158 sofia_handle_sip_i_invite() IP 192.168.2.107 Rejected by acl domains Why does it happens? ( May be this is mistakes in a new version?) Even if we have been change standart scenario SIPP to a call form a user "1000" $ ./sipp -sd uac > uac.xml ... changing name "sipp" to a "1000" everywhere ... $ ./sipp -sf uac-1000.xml -s 2000 -m 1 192.168.2.107 -trace_msg - we have the same result. How do we can get the same result, that inti version 1.0 rc4?