search for: txjitter

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090922/eb5fb16c/attachment.htm

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

...ats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve these values in my MySQL CDR table in order to calculate a MOS value: "ssrc=592614191;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=20734;rlp=0;rtt=0.094000" codec used: g711a -- -- -- Marc LEURENT lftsy at leurent.eu

Hi All, While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in my dialplan. I had 2 sip extensions 555 and 666 and I called from 555 to 666, but unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI. Would you please let me know what is wrong with my dialplan and/or what else should be done to get the value of ${RTPAUDIOQOS}? Following is my dialplan context

...formation about that in sources from 1.8, only a short reference in 1.4. Channel variables like CHANNEL(rtpqos,audio,rxjitter) show only information about the local channel. So not really usefull. In some old version they seemed to have it changed from remote_jitter to rxjitter, local_jitter to txjitter and so on. Was not even documented. The 2 variables RTPAUDIOQOSBRIDGED and RTPAUDIOQOS show exactly the things i want, but all information is stored in one field so its not really usable because it looks ugly in CDR report and doesnt show packet loss in %. The following interesting variables a...

...000000000044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69 #1 0x000000000049948f in pbx_builtin_setvar_helper (chan=0x2aaac801a890, name=0x2aaab69395a8 "RTPAUDIOQOS", value=0x2aaac80ecf20 "ssrc=1967815032;themssrc=917073588;lp=61288;rxjitter=0.000165;rxcount=3668;txjitter=0.005142;txcount=1515;rlp=0;rtt=3.924000") at pbx.c:5825 #2 0x00002aaab6925a94 in handle_request_bye (p=0x2aaac80ba4e0, req=0x40255b10) from /usr/lib/asterisk/modules/chan_sip.so #3 0x00002aaab69291ca in handle_request (p=0x2aaac80ba4e0, req=0x40255b10, sin=0x40255b00, recount=0x4...

...SIP/1001-000000b3 Variable: MEETMESECS Value: 64 Uniqueid: 1384275118.914 [Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1001-000000b3 Variable: RTPAUDIOQOS Value: ssrc=797521620;themssrc=278781419;lp=0;rxjitter=0.000489;rxcount=3885;txjitter=0.000000;txcount=3862;rlp=0;rtt=0.000000 Uniqueid: 1384275118.914 [Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1001-000000b3 Variable: RTPAUDIOQOSJITTER Value: minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxj...

...39; fax session '1', [ 038.968168 ], channel sent 2 frames (40 ms) of energy. [Apr 1 11:13:16] -- Executing [h at macro-redirection:1] Set("SIP/access-trunk-00000001", "CDR(userfield)= Hangup=16 RTCP=ssrc=1981386606;themssrc=2116388711;lp=0;rxjitter=0.000000;rxcount=2254;txjitter=0.000314;txcount=1949;rlp=0;rtt=0.000000") in new stack [Apr 1 11:13:16] -- Executing [h at macro-redirection:2] Hangup("SIP/access-trunk-00000001", "") in new stack [Apr 1 11:13:16] == Spawn extension (macro-redirection, h, 2) exited non-zero on 'SIP/access-tru...

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an