similar to: timing/clock problem

Hi, I'd like to have the logins that the accounts do, saved in a *sql table, with login name | IP | datetime, for audit porposes. How could I achieve that, if possible? Thanks in advanced, -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://dovecot.org/pipermail/dovecot/attachments/20180112/10e98846/attachment.html>

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Dear All, I am using Asterisk CVS-HEAD-05/05/05-15:46:09, and found it using up 99% of cpu 6-8 times in a day, even when its doing nothing or even if it is, its not supposed to eat all the cpu. whenever this happens, I need to stop it by issueing "stop now" 15-20 times on CLI or kill the process, then start over. Please suggest what to do. regards, Umair bari

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

Dears friends, I have been translating and extended, with educational porposes only, the available material in the www (http://zoonek2.free.fr/UNIX/48_R/all.html) "Statistcs avec R" for the Portuguese of Brazil. I consider the material very good and would like to share this work with anothers users R. Already I translated and I extended two chapters: Introduction to the R and analysis

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p><br></p><blockquote type="cite">On January 12, 2018 at 5:41 PM Alex JOST wrote:<br><br><br>Am 12.01.2018 um 16:23 schrieb Jorge Bastos:<blockquote type="cite">Hi,<br><br> <br><br>I'd

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First question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B (I have 2 computers with the same setup), will I be able to use SIP? I already have the two computers calling each other using IAX, but I would like to see if I can get them to work using SIP. My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been changed). After you've done that - try AGX again. HTH -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: 11 March 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users]

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

I've to project and build a fresh new box with * on. Basically, i'll have this situation: The office is connected to the phone carrier with a PRI. I need to let users continue to use their analog phones in a office, and an IP-phone based solution on the remote office, that will call using outside using the PRI on the first one. PRI---> Asterisk ---> Analog phones |

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then