On Sat, Nov 15, 2008 at 6:47 AM, Max Alex <[EMAIL PROTECTED]> wrote: > 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.
CLI> rtcp stats CLI> rtcp debug and as i recall you might also need "sip set debug on" in order to link this to calls/ip's, as rtcp stats are reporting only SIP call id. Regards, Atis > > Thanks, > Max Alex > Voip Developer > > > > On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen <[EMAIL PROTECTED]> > wrote: >> >> "Positively Optimistic" <[EMAIL PROTECTED]> 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 > > > _______________________________________________ > -- 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 > -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 _______________________________________________ -- 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