alok srivastava
2012-Jul-05 09:34 UTC
[asterisk-users] 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;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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120705/b13aa5d7/attachment.htm>
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 > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120705/8b051f9c/attachment.htm>