similar to: SNMP monitoring

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

Hi I'm trying to build the perl modules of net-snmp-5.4.2. Everything seems to work fine until "make test"[1]. The problem with "DynaLoader.pm" seems to be an old one. The machine is Centos 5.2 i386 I'm trying to build instead of using the one available in the repos because I'll build in other platforms and also in nagios-plugins. Thanks Marcelo [1]

Hi All, Twice now in the past few weeks I've walked into the office to find that our 1.2.24 Asterisk process is sat at 100%, and that hundreds of thousands of log files in /var/log/asterisk exist, all at 312 bytes, containing: Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Event Logger restarted Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Queue Logger restarted Aug 29 23:22:17

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

While attempting to package up the HEAD version of nut for our department/campus, I've run into a couple of minor issues with the latest HEAD - I'm going to give it another whack at a PW9170+ UPS and want to make sure I have the latest revision. :) First is, whilst building the drivers (SNMP required), I get: gmake[1]: Entering directory `/fsys2/source/nut-2.1/nut-2.1_cvs/drivers'

NUT developers, I'm looking for a low-tech solution for this linker error: > gcc -I../include -O -Wall -Wsign-compare -g -Dsolaris2 > -I/software/openssl-0.9.7_runtime/include -I. > -I/software/netsnmp-5.3/include -o snmp-ups snmp-ups.o main.o dstate.o \ > ../common/state.o ../common/upsconf.o ../common/parseconf.o \ > ../common/common.o ../common/setenv.o

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can

Hi, Can anyone tell me why the below dialplan doesn't filter off dialed numbers for 01793520158, and jump to "local",priority1 If I change it to : exten => 01793520158,1,Goto(local,${EXTEN:-3},1) .... then it works fine (but that's too specific)... exten => _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1) exten =>

Hi, I'm trying to rsync up to some centos repositories, but I only want to pull down the i386 and i386_64 folders with their RPMs, I've tried various combinations and include and exclude, and I'm sure that the below should work, but it doesn't... SOURCE=rsync://mirror.stanford.edu/mirrors/centos rsync -avrt $SOURCE --include=i386/ --include=*/ --exclude=* /var/www/html/centos/

I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box running Debian Sarge. res_snmp says its dependencies are netsnmp but Debian doesn't seem to have a netsnmp package. I've tried installing pretty much every package available related to snmp and no luck. I'm just wondering if anyone has successfully built the res_snmp module under Debian Sarge stable. Any

Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting "Got SIP response 489 "Bad event" back from 192.168.3.10" No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log

Hi All, I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K server behind NAT), and trying to pickup voicemail using Zoiper.. I can access the VM system, I hear all the prompts, and I can even hear part of the message playback. But then I get silence on the call (call stays up), and I get: Parsing

Hi All, I'd like to know if anyone has figured out a way to be able to have users logon/logoff manually from Cisco 79xx phones (with SIP firmware loaded)? Scenario is, user walks into office, sits at a random desk, and logs onto the phone. The system would need to "log them off" of the last hardphone they were on, and then configure the new phone for their extension. We're

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide:

Hi, I'm having complaints from some users about calls into dynamic meetme sessions failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug => debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function

Hi, I have problems compiling asterisk 1.4.11 with res_snmp. I do 'make menuselect', and I see that this resource module depends on netsnmp. I am using centOS 4.5. I do: > yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs I don't know if i am missing something. I go to the source directory and I do: ./configure but still does not work: > ... > checking for

Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn't find any clues. Someone find a way to use it? Thanks, Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070122/f28cdeac/attachment.htm

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have an NETYS RT 1/1 UPS, which is not fully supported. Starting snmp driver displays: [root at kvm1 ~]# snmp-ups -a local Network UPS Tools - Generic SNMP UPS driver 0.66 (2.6.4) Duplicate driver instance detected! Terminating other driver! No matching MIB found for sysOID '.1.3.6.1.4.1.2254.2.4'! Please report it to NUT developers, with an 'upsc' output for your