[Asterisk-Users] newbie debugger needs a little guidance

[phoneserver]

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:jtuthill at sip.jnctn.net>;tag=as58d6dd22
To:
<sip:jtuthill at sip.jnctn.net>;tag=1835cbfecbeb5b3c6b80319fb44e3d9b.f68f
Call-ID: 071f71b853555b2e3d2bb04915832c14 at 127.0.0.1
CSeq: 120 REGISTER
Contact: <sip:s at 192.168.3.70:5060>;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)