search for: anr

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. There are autocorrelations, so I model...

...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 Hold Last Message ip_peer (None) 58139462bde 00101/20006 0x0 (nothing) No Rx: REGISTER 1 active SIP channel Core show channels shows 0 active channels. Sip show channels shows 1 active channel. I find it odd to have 1 active c...

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 00000ms 0000ms 2 150.187.xxx.yyy ildefonso C72ACD25-1A 00101/11482 00000ms 0000ms 2 2 active SIP channel(s) -- SIP/fwd-161b answered SIP/ildefonso-d2fc -- Attempting nativ...

...achine 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 Hold Last Message 10.9.9.9 202 11a6323d2ff 00102/00000 0x100 (g729) No Tx: ACK 10.9.9.10 201 a8749-c0a80 00101/00001 0x100 (g729) No Rx: ACK Installing the open source binary of g279 (codec_g729.so in module...

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 Rx: PUBLISH...

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

...P/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2 <mailto:2944093@one_start:2> 2944093@one_start:2 Up Dial(SIP/2944093|36|tr) 2 active channels 1 active call hestia*CLI> hestia*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message xxx.yyy.128.115 (None) e77bba33-cc 00101/02261 unkn No Rx: REGISTER xxx.yyy.128.110 (None) 739f4603-e8 00101/00778 unkn No Rx: REGISTER xxx.yyy.128.86 (None) 56caad3a-eb 00101/01046 unkn No...

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 0221e874158bb62 0x4 (ulaw) No Tx: ACK SIPtrunkNu *.1*.19.70 (None) 2021549013484-1 0x0 (nothing) No Rx: OPTIONS...

...db: ldb_trace_next_request: (schema_load)->search ldb: ldb_trace_next_request: (lazy_commit)->search ldb: ldb_trace_next_request: (dirsync)->search ldb: ldb_trace_next_request: (paged_results)->search ldb: ldb_trace_next_request: (ranged_results)->search ldb: ldb_trace_next_request: (anr)->search ldb: ldb_trace_next_request: (server_sort)->search ldb: ldb_trace_next_request: (asq)->search ldb: ldb_trace_next_request: (extended_dn_in)->search ldb: ldb_trace_next_request: (descriptor)->search ldb: ldb_trace_next_request: (acl)->search ldb: ldb_trace_next_request: (a...

...uation 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 Seq (Tx/Rx) Form Hold Last Message xxx.187.128.20 3254104 06b217722a8 00102/00031 ulaw Yes Rx: REFER xxx.187.142.233 3254102 e1acc8e-eb8 00101/00002 ulaw No Rx: ACK 2 active SIP channels hermes*CLI> hermes*CLI> Mar 28 16:01:...

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 on...

...t;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 User/ANR Call ID Format Hold Last Message Expiry Peer >192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No Init: INVITE 0049351222 >192.168.200.10 00493511111111 1481837b-c0a801 0x4 (ulaw) No Rx: INVITE...

...m 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 Noise Reduction (ANR) - Bi-directional Comfort Noise Generator (CNG) - Double-talk/Coherence/Howling detection My question is whether the algorithm supports all the above functionalities ? Looking at speex_preprocess.h file, I find there are controls available for AGC, Noise Suppression, Echo Suppressio...

...7+3 8+4 9 10 [3,] 11 12 13 14 15 [4,] 16 17 18 19 20 [5,] 21 22 23 24 25 Is there an easy way of doing this, apart from iterating through the matrix d? Also, when I use > apply(d, c(1,2), function(x) {}) is there a way of knowing the column anr row which the element x is from, or o I have to use for loops for that? Thanks in advance, Rainer

...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 CVS-07/27/03-00:59:20 built by root@maui on a i686 running Linux maui*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 0 active SIP channel(s) maui*CLI> g.729 show license usage maui*CLI> There are currently no licenses of G.729 codec in use Asterisk Ready. -- x=0, open writing: /var/spool/asterisk/voicemail/default/1001/INBOX/msg0005 format: WAV, 0x...

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 Pri ) State...

...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 (Tx/Rx) Format 172.16.5.126 (None) 7167456b-51 00101/00621 UNKN 172.16.5.126 (None) fd6a6881-ea 00101/00621 UNKN 2 active SIP channel(s) fs-1*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active chan...

...ning 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 00102/00000 ulaw 10.0.0.48 3035 0008a3d2-05 00101/00102 ulaw 10.0.0.48 3035 0feb1c11386 00103/00101 g729 65.72.107.2 5126800422 28D20837-69 00103/00101 g729 4 active SIP chann...

...s 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 (Tx/Rx) Format 192.168.3.2 3018 2168101b16b 00102/00000 g729 I tried to find some old messages about this error but I couldn't find any clue. Any ideas?