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 <alok...@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@122.160.154.189>;tag=5a1cc54c > To: "2002" <sip:2002@122.160.154.189>;tag=as64f1f102 > 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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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