[asterisk-users] sip set debug on always showing error

[alok srivastava]

dear


please Help. I am continously getting this message after "sip set debug
on". and not getting clear voice from both side.


<--- Transmitting (NAT) to 122.163.193.94:1893 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.106:5060
;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893
From: "2002" <sip:2002 at 122.160.154.189>;tag=5a1cc54c
To: "2002" <sip:2002 at 122.160.154.189>;tag=as64f1f102
Call-ID: 8c18bd84585128a3f0885f54dfa966ba at 0:0:0:0:0:0:0:0
CSeq: 245 OPTIONS
Server: Asterisk PBX 10.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'8c18bd84585128a3f0885f54dfa966ba at 0:0:0:0:0:0:0:0'
in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84 at 0:0:0:0:0:0:0:0'
Method: OPTIONS
Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b at 0:0:0:0:0:0:0:0'
Method: OPTIONS
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