search for: themssrc

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/aste...

...something... I'm saving the SIP RTCP stats 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

...if (name[0] == '_') { (gdb) bt #0 0x000000000044da80 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=0x2a...

...event: Event: VarSet Privilege: dialplan,all Channel: 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=0.00...

...'Local/0XXXXXX40 at x-xxx-companyXXXXX-00000001;2' 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-r...

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