search for: rtpaudioqos

Displaying 17 results from an estimated 17 matches for "rtpaudioqos".

2009 Oct 01
2
help on ${RTPAUDIOQOS}
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 ${RTPAUDIOQO...
2009 Sep 22
3
RTPAUDIOQOS
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 ---------...
2014 Jan 11
0
RTPAUDIOQOS - Depending on who hangs up the phone, it's empty
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...
2009 Sep 10
1
RTPAUDIOQOS On DAHDI is it possible
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
2014 Jan 14
2
Asterisk QOS
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 S...
2011 Jan 28
1
CDR issue - Problem logging CDR(userfield) in Master.csv
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(billse...
2009 Aug 27
6
Measuring voice quality with Asterisk
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
2007 Aug 03
6
Measuring Jitter in Asterisk
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
2013 Dec 02
1
Not able to get remote channel variables containing RTCP values
...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 h...
2009 Feb 21
1
VoIP Information in CDRs
...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,...
2015 Mar 29
0
Iax2 statistics in dialplan
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
2006 Dec 08
1
SIP Quality Metrics
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2007 Nov 05
0
crash
...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/modul...
2013 Nov 12
1
Asterisk 1.8.20 crashing
...] 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: RTPAUDIOQO...
2009 Feb 25
1
Stuck Parked Calls?
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
2009 Aug 25
6
Breaking news, but what happened? 11.000 channels on one server
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,
2015 Apr 08
1
Help debugging a possible SIP channel leak in 11.17.0, possible race condition
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