Sip Set Debug On Always Showing Error

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;received2.163.193.94;rport93
From: “2002” ;tagZ1cc54c
To: “2002” ;tag=a64f1f102
Call-ID: 8c18bd84585128a3f0885f54dfa966ba@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@0:0:0:0:0:0:0:0′
in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0′
Method: OPTIONS
Really destroying SIP dialog ‘4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0′
Method: OPTIONS

2 Responses to “Sip Set Debug On Always Showing Error”

  1. SamyGo said:

    Jul 05, 12 at 5:08 am

    Hi,
    *CSeq: 245 OPTIONS *
    *
    *
    This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters.

    BR
    Sammy

  2. alok srivastava said:

    Jul 05, 12 at 7:15 am

    previously i was using for codec allow=al after that i changed disallow=al allow=silk24

    and i also change softph x-lite from jitsi(because of codec)
    now voice was coming fine from both side.
    But when i came to home from office not getting voice from both side. Threr is Airtel Broadband at my place.