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, εΧΗΕΞΙΚ ϊΟΜΟΤΟΧ 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.




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