similar to: Display variables

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, 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 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, 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 all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

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'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, 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'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, 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, 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 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 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 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

Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades

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:

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Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I