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

[SamyGo]

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


On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava <alokkic at gmail.com> wrote:

> 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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