similar to: Repeated: Got SIP response 489 "Bad event" back from

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 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

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/

Hi All, I've been reading up on 1.4 snmp integration. When I try and compile asterisk with a -with-netsnmp option it complains about net-snmp installation being broken. However, the net-snmp-devel rpm is installed, and snmpd on the machine runs fine. Anyone have a guide for the pre-requisites needed ? Cheers, Adrian -------------- next part -------------- An HTML attachment

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 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

Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see "Got SIP response 405 "Method Not Allowed" back from 192.168.3.64" but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat

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:

Hi, I used to have a set of samba shares working fine on a Centos 4 machine, accessed by XP clients and authenticated against a 2003 SBS server. Then I did a yum upgrade and a week later noticed that the seldom-used SMB shares have stopped working. No other changes to the Centos server, and no domain changes. The XP clients now just keep prompting for authentication. A wireshark trap shows

Also, it will be easier to provide helpful information if you'd describe what in your data you want to compare and what you hope to get out of the comparison. Best wishes, Ulrik Eric Berger <ericjberger at gmail.com> schrieb am Sa., 27. Jan. 2018, 08:18: > Hi Marsh, > An RDS is not a data structure such as a data.frame. It can be anything. > For example if I want to save my

Hi, Thank you for taking a look at this. This is 13.10. I realize Saucy is no longer supported. This system is slated for replacement, but in the meantime I'm responsible for it. The repo version of nut was 2.6.4-2.3ubuntu2. Installing it was useful as it gave me the nut user and group. The compiled version errors out whether or not the repo version is installed. I just did strace.

If your two objects have class "data.frame" (look at class(objectName)) and they both have the same number of columns and the same order of columns and the column types match closely enough (use all.equal(x1, x2) for that), then you can try which( rowSums( x1 != x2 ) > 0) E.g., > x1 <- data.frame(X=1:5, Y=rep(c("A","B"),c(3,2))) > x2 <-

The anti_join from the package dplyr might also be handy. install.package("dplyr") library(dplyr) anti_join (x1, x2) You can get help on the different functions by ?function.name(), so ?anti_join() will bring you help - and examples - on the anti_join function. It might be worth testing your approach on a small subset of the data. That makes it easier for you to follow what happens

Hi Guys, I apologize for my rank & utter newness at R. I used summary() and found about 95 variables, both character and numeric, all with "Length:368842" I assume is the # of records. I'd like to know the record number (row #?) of any record where the data doesn't match in the 2 files of what should be the same output. Thanks in advance, M. //