[asterisk-users] RTP LOG
Hi All, I am using asterisk 1.4.22 in my local system I want to know how can we set ability to log and report RTP and jitter statistics per call. Is there any configuration in logger or configuration in rtp? Please provide some guide lines for this. Thanks in advance! Thanks, Max Alex Voip Developer ___ -- 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
Re: [asterisk-users] RTP LOG
This works for us exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS}) exten = h,2,Hangup() results in Set(SIP/rpx2399a-b61fc5e0, CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00) *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Max Alex *Sent:* Friday, November 14, 2008 5:25 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED]; [EMAIL PROTECTED] *Subject:* [asterisk-users] RTP LOG Hi All, I am using asterisk 1.4.22 in my local system I want to know how can we set ability to log and report RTP and jitter statistics per call. Is there any configuration in logger or configuration in rtp? Please provide some guide lines for this. Thanks in advance! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP LOG
Positively Optimistic [EMAIL PROTECTED] writes: exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS}) exten = h,2,Hangup() results in Set(SIP/rpx2399a-b61fc5e0, CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00) 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
Re: [asterisk-users] RTP LOG
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 [EMAIL PROTECTED][EMAIL PROTECTED] wrote: Positively Optimistic [EMAIL PROTECTED] writes: exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS}) exten = h,2,Hangup() results in Set(SIP/rpx2399a-b61fc5e0, CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00) Does it still only report what was in the last incoming RTCP packet? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP LOG
On Sat, Nov 15, 2008 at 6:47 AM, Max Alex [EMAIL PROTECTED] wrote: Hi All, Thanks for reply i have tried for this, it looks fine for me, but is there any way to check rtp log while call is connected or any way to enable it to write in log file. Please give me some guide lines! thanks in advance. CLI rtcp stats CLI rtcp debug and as i recall you might also need sip set debug on in order to link this to calls/ip's, as rtcp stats are reporting only SIP call id. Regards, Atis Thanks, Max Alex Voip Developer On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen [EMAIL PROTECTED] wrote: Positively Optimistic [EMAIL PROTECTED] writes: exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS}) exten = h,2,Hangup() results in Set(SIP/rpx2399a-b61fc5e0, CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00) 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