[asterisk-users] RTP LOG

Max Alex max.asterisk at gmail.com
Fri Nov 14 22:47:42 CST 2008


Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in advance.

Thanks,
Max Alex
Voip Developer



On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen
<benny+usenet at amorsen.dk<benny%2Busenet at amorsen.dk>
> wrote:

> "Positively Optimistic" <positivelyoptimistic at gmail.com> writes:
>
> > exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
> > exten => h,2,Hangup()
> > results in....
> > Set("SIP/rpx2399a-b61fc5e0",
> >
> "CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000")
>
> Does it still only report what was in the last incoming RTCP packet?
>
>
> /Benny
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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/20081115/ce6b4431/attachment.htm 


More information about the asterisk-users mailing list