search for: rtpaudioqo

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 ${RTPAUDIOQ...

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 --------...

I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up. Any idea why I would see this? exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS}) Thanks, Richar...

hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... Any Idea..!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/810a0848/attachment.htm

I asked this on the list over the weekend, and likely missed a few people inboxes. I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up. Any idea why I would see this? exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS}) Thanks, Richard...

Dear all, I am having an issue with CDR logging. What I want to do is log jitter variable from RTPAUDIOQOS module into Master.csv at the end of each call. I am using asterisk version 1.4.26. For CDR purposes, I am using cdr_custom, and the content of my cdr_custom.conf is the following: [mappings] Master.csv => ${CDR(dstchannel)},${CDR(clid)},${CDR(cid-num)},${CDR(dst)},${CDR(start)},${CDR(bills...

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

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

...nce 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 are completely empty (show 0), here is how i write it to CDR in...

...he peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)} Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(audionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS}) The problems I have so far: *1. CODEC *Codec is reported only for A-Leg. When transcoding asterisk logs the above line as: slin for read / slin for write / the codec of A-Leg / 0 for t.38. Is there a way to get the codec for both legs of a call? *2. RTP Qos is reported only for A-Leg. *Also,...

Hi All How to have access to the IAX2 call statistics inside the dialplan (not CLI)? I have no IAX2 clients (yet) to test, but do RTPAUDIOQOS.* variables do the job? Are they available to IAX2 calls as they are for SIP? Stats like total packets sent and received, lost pkts, rtt, etc. would be nice. cheers Ethy

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...00044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69 69 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/modu...

...] DEBUG[23332] manager.c: Examining 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: RTPAUDIOQ...

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

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

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