Displaying 5 results from an estimated 5 matches for "local_jitter".
2011 Sep 05
1
Variables error in 1.8.6.0.
...n => 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, audio, local_jitter)})
; the Same for jitter
Theoretically this should throw these variables in a table in MySQL but
these values ??cannot be readed. I think it's a different syntax in 1.8.
I gave this error:
- Executing [h @ macro-special1: 11] Set ("SIP/1010-00000002",...
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.
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2013 Dec 02
1
Not able to get remote channel variables containing RTCP values
...ant
find any information 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...
2011 Jul 03
1
SIP Peer Name Variable
Hi,
Is there a variable that contains the Sip Peer name?
I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else.
I need a variable that is always set to the SIP Peer's name.
Thanks
Dan
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2011 Oct 11
11
Reporting for Asterisk Call Center
Dear Tariq;
About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting?
Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).