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

[alok srivastava]

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 <govoiper at 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 <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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>
>
> --
> _____________________________________________________________________
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