similar to: help on ${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 -------------- next part

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'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 Seguin -------------- next part

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

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 =>

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve 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

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

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 =>

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,

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the 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)}

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

Hi all, I have seen a lot of message talking about asterisk crashed when using queue and mixmonitor together. I do use both in our system and also get the crash (segfault) randomly. I don't know it is related to the reason above as I have no idea about how it happened. I get the core dump below. If anybody has any idea about the root cause of the crash, please tell me. Asterisk 1.4.13

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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Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk,

I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says "Packet2Packet Bridding SIP/blah to SIP/blah". I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in