[Freeswitch-users] How to configure FreeSWITCH for calling SIPprofile "internal"?

Евгений Золотов zolotov at altron.ua
Tue Sep 2 05:30:27 PDT 2008


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 at 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 at 127.0.0.1:5061>;tag=5239SIPpTag0079
To: sut <sip:2000 at 192.168.2.107:5060>;tag=3KKN20eKpFFrN
Call-ID: 79-5239 at 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 at 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 at 127.0.0.1:5061>;tag=5566SIPpTag001
To: sut <sip:2000 at 192.168.2.107:5060>;tag=2Qv35F85NtN9D
Call-ID: 1-5566 at 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 at 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 at freeswitch.org














------------------------------------------------------------------------------


  _______________________________________________
  Freeswitch-users mailing list
  Freeswitch-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080902/343b9edd/attachment-0002.html 


More information about the FreeSWITCH-users mailing list