previously i was using for codec allow=all after that i changed disallow=all 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. On Thu, Jul 5, 2012 at 3:36 PM, SamyGo <govoi...@gmail.com> wrote: > 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 >> > > > -- > _____________________________________________________________________ > -- 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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