you really should update to 4.4 as there are better options to address some ITSP's who want different options.
In your case I "think" 4.2 has the option to change this here. Use default asserted identity(Default: checked)The asserted identity header may be used by the ITSP to determine the originating party for call setup requests. If checked (default), use the default asserted identity. Otherwise, you must enter a username@domain to override the default. Asserted identityOverride the default asserted identity. On Thu, Feb 23, 2012 at 10:53 AM, Thomas <[email protected]> wrote: > > Hello, > > we are using sipxecs 4.2.1 and are now trying to enable a SIP trunk. > > But we are rejected from our SIP provider, because the wrong TO: > domain is sent by sipxecs > > To:<sip:[email protected];user=phone > > should be > > To:<sip:[email protected] > > and not @sipxecs.sos-kd.org - sipxecs.sos-kd.org is the host name of > sipxecs itself. > > Any advice would be really welcome, > > Thanks, > Thomas > > > > "2012-02-23T14:35:32.935000Z":357:JAVA:DEBUG:sipxecs.sos-kd.org:Timer-1:00000000:SipListenerImpl:"DialogCreated > by request: INVITE > sip:[email protected]:5060;user=phone;transport=udp;sipxecs-lineid=8 > SIP/2.0\r\nRoute: <sip:a.b.c.d:5090;lr>\r\nRecord-Route: > <sip:a.b.c.d:5060;lr;sipXecs-CallDest=CUST;sipXecs-rs=%2Aauth%7E.%2Afrom%7ENjhjNjNjZjZkMw%60%60.900_ntap%2Aid%7EMTExOTctMA%60%60%217607ef9250b96a03deca90c6725ca92c>\r\nFrom: > \"Kofler Thomas\" > <sip:[email protected];user=phone>;epid=6AF60C3424;tag=68c63cf6d3\r\nTo: > <sip:[email protected];user=phone>\r\nCSeq: 163 > INVITE\r\nCall-ID: 246c39fa-72b2-4f85-8177-1995a664cb9d\r\nMax-Forwards: > 18\r\nVia: SIP/2.0/TCP > a.b.c.d;branch=z9hG4bK-XX-000avCs7YpVIPNuXT2IDxNgNow;rport=45512\r\nVia: > SIP/2.0/UDP > a.b.c.d;branch=z9hG4bK-XX-0007KG2Wft9wispodY5dnKdZXQ~070nO800F3VT9Wa1Yqf0iw;id=11197-0\r\nVia: > SIP/2.0/TCP w.x.y.z:55332;branch=z9hG4bK7021a177\r\nContact: > <sip:INNKDI303App.kd.sos:5060;transpor > t=Tcp;maddr=w.x.y.z;ms-opaque=bfc793fad807176c;x-sipX-nonat>\r\nSupported: > 100rel\r\nUser-Agent: RTCC/4.0.0.0 MediationServer\r\nContent-Type: > application/sdp\r\nAllow: ACK,CANCEL,BYE,INVITE,PRACK,UPDATE\r\nDate: Thu, > 23 Feb 2012 14:35:24 GMT\r\nExpires: 7200\r\nX-Sipx-Handled: > Xa.b.c.d-195.190.143.104\r\nContent-Length: 364\r\n\r\nv=0\r\no=- 23 1 IN > IP4 w.x.y.z\r\ns=session\r\nc=IN IP4 w.x.y.z\r\nb=CT:1000\r\nt=0 > 0\r\nm=audio 30248 RTP/AVP 97 101 13 0 8\r\nc=IN IP4 > a.b.c.d\r\na=label:Audio\r\na=sendrecv\r\na=rtpmap:97 > RED/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 > 0-16\r\na=rtpmap:13 CN/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 > PCMA/8000\r\na=ptime:20\r\na=x-sipx-ntap:Xa.b.c.d-195.190.143.104;0\r\n" > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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