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Dear list, I am trying to re-analyse something. I do have two time series, one of which (ts.mar) might help explaining the other (ts.anr). In the original analysis, no-one seems to have cared about the data being time-series and they just did OLS. This yielded a strong positive correlation. I want to know if this correlation is still as strong when the autocorrelations are taken into account.

vps*CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 0 active IAX channels vps*CLI> core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi, I'm wondering if this is normal asterisk behaviour: asterisk*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 10.12.0.2 (None) 3c2f7ff2975e-wp 0x0 (nothing) No Rx: PUBLISH <guest> 10.12.0.2 (None) 3c2f7f21b71b-9q 0x0 (nothing) No

Hello all, ? this is my first Post in a Mailing List I hope everything goes fine. ? We are running a Samba 4 DC (4.0.14, Version 4.1.4 has the same problem) as a second DC in our Windows Environment. This server is in a second site. ? So after a while Samba 4 hangs and it is not possible to talk to the server via the RPC Protocol. So all samba-tools Commands like ?samba-tool drs showrepl? run

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

Hi. I'm trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI. I send the command "sip show channels", and I get back a response along the lines of (* used to protect the innocent...): Peer User/ANR Call ID Format Hold Last Message Expiry Peer *8.22.*0.34 02035644444

Hello. Samba 4.1.0pre1-GIT-aad669b, joined as a DC to an existing domain. At least 6 accounts behave like this: Kerberos: AS-REQ techgroup at KLIN.KIFATO-MK.COM from ipv4:192.168.1.31:33822 for krbtgt/KLIN.KIFATO-MK.COM at KLIN.KIFATO-MK.COM ldb: ldb_trace_request: SEARCH dn: <rootDSE> scope: sub expr: (&(objectClass=user)(userPrincipalName=techgroup at KLIN.KIFATO-MK.COM)) control:

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

ANR is a international news station we were testing on icecast over the weekend the quality great we chose mp3 because anyone can hear it. linux, mac. or windblows. also wanted to use a linux server, which is far more reliable than a windows machine ( always dropping out for some reason ) If the ogg only rule is permanent we will have to talk very nicely to our system admin to switch to ogg with a

Hi Luca Y need to check your wifes codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer

Hi, I am just starting up on SPEEX AEC algorithm and have couple of questions around echo cancelation algorithm used in SPEEX. 1) Usually Echo Cancelation Algorithm has support for number of components ? - Non-Linear Processor (NLP) - Automatic Microphone Gain Control (AGC) - Transducer Equalization (EQ) - Dynamic Range Compression (DRC) - Ambient

Hi I have another question concerning matrices: I have two matrices: > b <- matrix(1:25,5,5,byrow=T) > b [,1] [,2] [,3] [,4] [,5] [1,] 1 2 3 4 5 [2,] 6 7 8 9 10 [3,] 11 12 13 14 15 [4,] 16 17 18 19 20 [5,] 21 22 23 24 25 and > d <- matrix(1:4,2,2,byrow=T) > d [,1] [,2] [1,] 1 2 [2,] 3 4

Upon testing the g.729 licenses in our lab we found the following issues in a plain SIP environment: 1. When g.729 licenses are used in leaving a Voicemail, they do not get released upon clearing the channel. 2. If the voicemail.conf has options for 2 format types (ie. wav49 and WAV), it consumes 2 g.729 licenses. I have reported this to the bugtracker. maui*CLI> show version Asterisk

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

I'm having an oddball issue with a Polycom SoundPoint IP 500. As you can see below Asterisk thinks there are 2 SIP channels active, but show channels tells me there are no calls active. Anyone have any idea why this is happening? The Polycom occasionally stops accepting calls and requires a power cycle. fs-1*CLI> sip show channels Peer User/ANR Call ID Seq

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq