Hi guys, I am trying to step our asterisk server. All the internal phones / extensions work and I had the outgoing / incoming calls working before. But for some reason, unknown to me, it has stopped working.
I have switched on sip debug and the main thing I notice is the recurring appearance of "Noisy feedback tells: pid=2359 req_src_ip=217.155.69.86 req_src_port=5060 in_uri=sip:sip.jnctn.net out_uri=sip:sip.jnctn.net via_cnt==1" Can anyone help me with this? Thanks, James p.s. Here is a bit of the console debug output. Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK0438ec30 From: <sip:[EMAIL PROTECTED]>;tag=as58d6dd22 To: <sip:[EMAIL PROTECTED]>;tag=1835cbfecbeb5b3c6b80319fb44e3d9b.f68f Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060>;expires=120 Server: OpenSer (1.0.0-pre0 (i386/linux)) Content-Length: 0 Warning: 392 66.227.100.20:5060 "Noisy feedback tells: pid=2359 req_src_ip=217.155.69.86 req_src_port=5060 in_uri=sip:sip.jnctn.net out_uri=sip:sip.jnctn.net via_cnt==1" 10 headers, 0 lines Feb 28 07:20:09 NOTICE[8591]: chan_sip.c:6831 handle_response: Outbound Registration: Expiry for sip.jnctn.net is 120 sec (Scheduling reregistration in 105000 ms) _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
